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[PATCH 0/3] ASoC: SOF: ipc4: Add core_id support from topology
by Peter Ujfalusi 08 Mar '23

08 Mar '23

Hi, The following series will add support for handling the core_id token which is needed for supporting multiple cores with complex topologies. Regards, Peter --- Peter Ujfalusi (2): ASoC: SOF: ipc4: Add macro to set the core_id in create_pipe message ASoC: SOF: ipc4-topology: Add support for core_id for pipelines Rander Wang (1): ASoC: SOF: ipc4-topology: add core token in each module extended token list include/sound/sof/ipc4/header.h | 4 ++++ sound/soc/sof/ipc4-topology.c | 13 ++++++++++--- sound/soc/sof/ipc4-topology.h | 2 ++ 3 files changed, 16 insertions(+), 3 deletions(-) -- 2.39.2

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[PATCH] ASoC: SOF: ipc3: Check for upper size limit for the received message
by Peter Ujfalusi 08 Mar '23

08 Mar '23

The sof_ipc3_rx_msg() checks for minimum size of a new rx message but it is missing the check for upper limit. Corrupted or compromised firmware might be able to take advantage of this to cause out of bounds reads outside of the message area. Reported-by: Curtis Malainey <cujomalainey(a)chromium.org> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Curtis Malainey <curtis(a)malainey.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/ipc3.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/ipc3.c b/sound/soc/sof/ipc3.c index 3de64ea2dc9a..4493bbd7faf1 100644 --- a/sound/soc/sof/ipc3.c +++ b/sound/soc/sof/ipc3.c @@ -970,8 +970,9 @@ static void sof_ipc3_rx_msg(struct snd_sof_dev *sdev) return; } - if (hdr.size < sizeof(hdr)) { - dev_err(sdev->dev, "The received message size is invalid\n"); + if (hdr.size < sizeof(hdr) || hdr.size > SOF_IPC_MSG_MAX_SIZE) { + dev_err(sdev->dev, "The received message size is invalid: %u\n", + hdr.size); return; } -- 2.39.2

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[PATCH] ASoC: SOF: topology: Fix error handling in sof_widget_ready()
by Peter Ujfalusi 08 Mar '23

08 Mar '23

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Fix the error paths in sof_widget_ready() to free all allocated memory and prevent memory leaks. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/topology.c | 34 +++++++++++++++++----------------- 1 file changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 4a62ccc71fcb..9f3a038fe21a 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1388,14 +1388,15 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, if (ret < 0) { dev_err(scomp->dev, "failed to parse component pin tokens for %s\n", w->name); - return ret; + goto widget_free; } if (swidget->num_sink_pins > SOF_WIDGET_MAX_NUM_PINS || swidget->num_source_pins > SOF_WIDGET_MAX_NUM_PINS) { dev_err(scomp->dev, "invalid pins for %s: [sink: %d, src: %d]\n", swidget->widget->name, swidget->num_sink_pins, swidget->num_source_pins); - return -EINVAL; + ret = -EINVAL; + goto widget_free; } if (swidget->num_sink_pins > 1) { @@ -1404,7 +1405,7 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, if (ret < 0) { dev_err(scomp->dev, "failed to parse sink pin binding for %s\n", w->name); - return ret; + goto widget_free; } } @@ -1414,7 +1415,7 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, if (ret < 0) { dev_err(scomp->dev, "failed to parse source pin binding for %s\n", w->name); - return ret; + goto widget_free; } } @@ -1436,9 +1437,8 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, case snd_soc_dapm_dai_out: dai = kzalloc(sizeof(*dai), GFP_KERNEL); if (!dai) { - kfree(swidget); - return -ENOMEM; - + ret = -ENOMEM; + goto widget_free; } ret = sof_widget_parse_tokens(scomp, swidget, tw, token_list, token_list_size); @@ -1496,8 +1496,7 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, tw->shift, swidget->id, tw->name, strnlen(tw->sname, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) > 0 ? tw->sname : "none"); - kfree(swidget); - return ret; + goto widget_free; } if (sof_debug_check_flag(SOF_DBG_DISABLE_MULTICORE)) { @@ -1518,10 +1517,7 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, if (ret) { dev_err(scomp->dev, "widget event binding failed for %s\n", swidget->widget->name); - kfree(swidget->private); - kfree(swidget->tuples); - kfree(swidget); - return ret; + goto free; } } } @@ -1532,10 +1528,8 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, spipe = kzalloc(sizeof(*spipe), GFP_KERNEL); if (!spipe) { - kfree(swidget->private); - kfree(swidget->tuples); - kfree(swidget); - return -ENOMEM; + ret = -ENOMEM; + goto free; } spipe->pipe_widget = swidget; @@ -1546,6 +1540,12 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, w->dobj.private = swidget; list_add(&swidget->list, &sdev->widget_list); return ret; +free: + kfree(swidget->private); + kfree(swidget->tuples); +widget_free: + kfree(swidget); + return ret; } static int sof_route_unload(struct snd_soc_component *scomp, -- 2.39.2

2 1

[PATCH] ASoC: Intel: soc-acpi: fix copy-paste issue in topology names
by Peter Ujfalusi 08 Mar '23

08 Mar '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> For some reason the convention for topology names was not followed and the name inspired by another unrelated hardware configuration. As a result, the kernel will request a non-existent topology file. Link: https://github.com/thesofproject/sof/pull/6878 Fixes: 2ec8b081d59f ("ASoC: Intel: soc-acpi: Add entry for sof_es8336 in ADL match table") Cc: stable(a)vger.kernel.org Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/intel/common/soc-acpi-intel-adl-match.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index 56ee5fef66a8..28dd2046e4ac 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -559,7 +559,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { { .comp_ids = &essx_83x6, .drv_name = "sof-essx8336", - .sof_tplg_filename = "sof-adl-es83x6", /* the tplg suffix is added at run time */ + .sof_tplg_filename = "sof-adl-es8336", /* the tplg suffix is added at run time */ .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_SSP_NUMBER | SND_SOC_ACPI_TPLG_INTEL_SSP_MSB | SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, -- 2.39.2

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[PATCH] ASoC: SOF: Intel: hda-dsp: harden D0i3 programming sequence
by Peter Ujfalusi 08 Mar '23

08 Mar '23

From: Rander Wang <rander.wang(a)intel.com> Add delay between set and wait command according to hardware programming sequence. Also add debug log to detect error. Signed-off-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/intel/hda-dsp.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index e3b69dbc1308..c9231aeacc53 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -392,6 +392,12 @@ static int hda_dsp_update_d0i3c_register(struct snd_sof_dev *sdev, u8 value) snd_sof_dsp_update8(sdev, HDA_DSP_HDA_BAR, chip->d0i3_offset, SOF_HDA_VS_D0I3C_I3, value); + /* + * The value written to the D0I3C::I3 bit may not be taken into account immediately. + * A delay is recommended before checking if D0I3C::CIP is cleared + */ + usleep_range(30, 40); + /* Wait for cmd in progress to be cleared before exiting the function */ ret = hda_dsp_wait_d0i3c_done(sdev); if (ret < 0) { @@ -400,6 +406,12 @@ static int hda_dsp_update_d0i3c_register(struct snd_sof_dev *sdev, u8 value) } reg = snd_sof_dsp_read8(sdev, HDA_DSP_HDA_BAR, chip->d0i3_offset); + /* Confirm d0i3 state changed with paranoia check */ + if ((reg ^ value) & SOF_HDA_VS_D0I3C_I3) { + dev_err(sdev->dev, "failed to update D0I3C!\n"); + return -EIO; + } + trace_sof_intel_D0I3C_updated(sdev, reg); return 0; -- 2.39.2

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[PATCH] ASoC: SOF: Intel: hda-ctrl: re-add sleep after entering and exiting reset
by Peter Ujfalusi 08 Mar '23

08 Mar '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> This reverts commit a09d82ce0a867 ("ASoC: SOF: Intel: hda-ctrl: remove useless sleep") It was a mistake to remove those delays, in light of comments in the HDaudio spec captured in snd_hdac_bus_reset_link() that the codec needs time for its initialization and PLL lock. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/intel/hda-ctrl.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c index a1037512da1f..e1dba6b79065 100644 --- a/sound/soc/sof/intel/hda-ctrl.c +++ b/sound/soc/sof/intel/hda-ctrl.c @@ -198,12 +198,15 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev) goto err; } + usleep_range(500, 1000); + /* exit HDA controller reset */ ret = hda_dsp_ctrl_link_reset(sdev, false); if (ret < 0) { dev_err(sdev->dev, "error: failed to exit HDA controller reset\n"); goto err; } + usleep_range(1000, 1200); hda_codec_detect_mask(sdev); -- 2.39.2

2 1

[PATCH] ASoC: SOF: Intel: pci-tng: revert invalid bar size setting
by Peter Ujfalusi 08 Mar '23

08 Mar '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The logic for the ioremap is to find the resource index 3 (IRAM) and infer the BAR address by subtracting the IRAM offset. The BAR size defined in hardware specifications is 2MB. The commit 5947b2726beb6 ("ASoC: SOF: Intel: Check the bar size before remapping") tried to find the BAR size by querying the resource length instead of a pre-canned value, but by requesting the size for index 3 it only gets the size of the IRAM. That's obviously wrong and prevents the probe from proceeding. This commit attempted to fix an issue in a fuzzing/simulated environment but created another on actual devices, so the best course of action is to revert that change. Reported-by: Ferry Toth <fntoth(a)gmail.com> Tested-by: Ferry Toth <fntoth(a)gmail.com> (Intel Edison-Arduino) Link: https://github.com/thesofproject/linux/issues/3901 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/intel/pci-tng.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/sof/intel/pci-tng.c b/sound/soc/sof/intel/pci-tng.c index 5b2b409752c5..8c22a00266c0 100644 --- a/sound/soc/sof/intel/pci-tng.c +++ b/sound/soc/sof/intel/pci-tng.c @@ -75,11 +75,7 @@ static int tangier_pci_probe(struct snd_sof_dev *sdev) /* LPE base */ base = pci_resource_start(pci, desc->resindex_lpe_base) - IRAM_OFFSET; - size = pci_resource_len(pci, desc->resindex_lpe_base); - if (size < PCI_BAR_SIZE) { - dev_err(sdev->dev, "error: I/O region is too small.\n"); - return -ENODEV; - } + size = PCI_BAR_SIZE; dev_dbg(sdev->dev, "LPE PHY base at 0x%x size 0x%x", base, size); sdev->bar[DSP_BAR] = devm_ioremap(sdev->dev, base, size); -- 2.39.2

2 1

[PATCH] ASoC: SOF: Intel: mtl: Access MTL_HFPWRCTL from HDA_DSP_BAR
by Peter Ujfalusi 08 Mar '23

08 Mar '23

From: Yong Zhi <yong.zhi(a)intel.com> The Host Power Management/Clock Control (ULP) Registers in the HDA BAR shadow the values of the same registers in the DSP BAR, so let's modify the latter - as done already for other accesses. Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/intel/mtl.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index 216fd07a3a93..8f0ed1cbf3b5 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -256,7 +256,7 @@ static int mtl_dsp_pre_fw_run(struct snd_sof_dev *sdev) dev_err(sdev->dev, "failed to power up gated DSP domain\n"); /* make sure SoundWire is not power-gated */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, MTL_HFPWRCTL, + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, MTL_HFPWRCTL, MTL_HfPWRCTL_WPIOXPG(1), MTL_HfPWRCTL_WPIOXPG(1)); return ret; } -- 2.39.2

2 1

[PATCH 0/4] SoC: SOF: Intel: FIx device descriptions (missing ops_free)
by Peter Ujfalusi 08 Mar '23

08 Mar '23

Hi, We have recently noticed that the ops_free callback was missed for the device descriptions on Intel platforms. Regards, Peter --- Ranjani Sridharan (4): ASoC: SOF: Intel: MTL: Fix the device description ASoC: SOF: Intel: HDA: Fix device description ASoC: SOF: Intel: SKL: Fix device description ASOC: SOF: Intel: pci-tgl: Fix device description sound/soc/sof/intel/pci-apl.c | 1 + sound/soc/sof/intel/pci-cnl.c | 2 ++ sound/soc/sof/intel/pci-icl.c | 1 + sound/soc/sof/intel/pci-mtl.c | 1 + sound/soc/sof/intel/pci-skl.c | 2 ++ sound/soc/sof/intel/pci-tgl.c | 7 +++++++ 6 files changed, 14 insertions(+) -- 2.39.2

2 5

[PATCH] ASoC: SOF: sof-audio: don't squelch errors in WIDGET_SETUP phase
by Peter Ujfalusi 08 Mar '23

08 Mar '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> When an IPC error happens while setting-up a widget during the FE hw_params phase, the existing logic will unwind all previous configurations but will overwrite the return status. The ALSA/ASoC logic will then proceed with the prepare and trigger phases, even though the firmware resources are not available. Fix by returning the initial error code and ignoring the code returned in the UNPREPARE phase. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Chao Song <chao.song(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/sof-audio.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index d7df29f2ada8..6de388a8d0b8 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -610,8 +610,8 @@ int sof_widget_list_setup(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm, ret = sof_walk_widgets_in_order(sdev, spcm, fe_params, platform_params, dir, SOF_WIDGET_SETUP); if (ret < 0) { - ret = sof_walk_widgets_in_order(sdev, spcm, fe_params, platform_params, - dir, SOF_WIDGET_UNPREPARE); + sof_walk_widgets_in_order(sdev, spcm, fe_params, platform_params, + dir, SOF_WIDGET_UNPREPARE); return ret; } -- 2.39.2

2 1

[PATCH] ASoC: SOF: sof-audio: Fix broken early bclk feature for SSP
by Peter Ujfalusi 08 Mar '23

08 Mar '23

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> With the removal of widget setup during BE hw_params, the DAI config IPC is never sent with the SOF_DAI_CONFIG_FLAGS_HW_PARAMS. This means that the early bit clock feature required for certain codecs will be broken. Fix this by saving the config flags sent during BE DAI hw_params and reusing it when the DAI_CONFIG IPC is sent after the DAI widget is set up. Also, free the DAI config before the widget is freed. The DAI_CONFIG IPC sent during the sof_widget_free() does not have the DAI index information. So, save the dai_index in the config during hw_params and reuse it during hw_free. For IPC4, do not clear the node ID during hw_free. It will be needed for freeing the group_ida during unprepare. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/ipc3-topology.c | 32 ++++++++++++++++++++++++++++++-- sound/soc/sof/ipc4-topology.c | 15 +++++++++++++-- sound/soc/sof/sof-audio.c | 28 +++++++++++++++++++++++++--- 3 files changed, 68 insertions(+), 7 deletions(-) diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index dceb78bfe17c..b1f425b39db9 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -2081,7 +2081,9 @@ static int sof_ipc3_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget * break; case SOF_DAI_INTEL_ALH: if (data) { - config->dai_index = data->dai_index; + /* save the dai_index during hw_params and reuse it for hw_free */ + if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) + config->dai_index = data->dai_index; config->alh.stream_id = data->dai_data; } break; @@ -2089,7 +2091,30 @@ static int sof_ipc3_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget * break; } - config->flags = flags; + /* + * The dai_config op is invoked several times and the flags argument varies as below: + * BE DAI hw_params: When the op is invoked during the BE DAI hw_params, flags contains + * SOF_DAI_CONFIG_FLAGS_HW_PARAMS along with quirks + * FE DAI hw_params: When invoked during FE DAI hw_params after the DAI widget has + * just been set up in the DSP, flags is set to SOF_DAI_CONFIG_FLAGS_HW_PARAMS with no + * quirks + * BE DAI trigger: When invoked during the BE DAI trigger, flags is set to + * SOF_DAI_CONFIG_FLAGS_PAUSE and contains no quirks + * BE DAI hw_free: When invoked during the BE DAI hw_free, flags is set to + * SOF_DAI_CONFIG_FLAGS_HW_FREE and contains no quirks + * FE DAI hw_free: When invoked during the FE DAI hw_free, flags is set to + * SOF_DAI_CONFIG_FLAGS_HW_FREE and contains no quirks + * + * The DAI_CONFIG IPC is sent to the DSP, only after the widget is set up during the FE + * DAI hw_params. But since the BE DAI hw_params precedes the FE DAI hw_params, the quirks + * need to be preserved when assigning the flags before sending the IPC. + * For the case of PAUSE/HW_FREE, since there are no quirks, flags can be used as is. + */ + + if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) + config->flags |= flags; + else + config->flags = flags; /* only send the IPC if the widget is set up in the DSP */ if (swidget->use_count > 0) { @@ -2097,6 +2122,9 @@ static int sof_ipc3_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget * &reply, sizeof(reply)); if (ret < 0) dev_err(sdev->dev, "Failed to set dai config for %s\n", dai->name); + + /* clear the flags once the IPC has been sent even if it fails */ + config->flags = SOF_DAI_CONFIG_FLAGS_NONE; } return ret; diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 3e27c7a48ebd..37927e9a700d 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -980,6 +980,7 @@ static void sof_ipc4_unprepare_copier_module(struct snd_sof_widget *swidget) ipc4_copier = dai->private; if (ipc4_copier->dai_type == SOF_DAI_INTEL_ALH) { + struct sof_ipc4_copier_data *copier_data = &ipc4_copier->data; struct sof_ipc4_alh_configuration_blob *blob; unsigned int group_id; @@ -989,6 +990,9 @@ static void sof_ipc4_unprepare_copier_module(struct snd_sof_widget *swidget) ALH_MULTI_GTW_BASE; ida_free(&alh_group_ida, group_id); } + + /* clear the node ID */ + copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK; } } @@ -1940,8 +1944,15 @@ static int sof_ipc4_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget * pipeline->skip_during_fe_trigger = true; fallthrough; case SOF_DAI_INTEL_ALH: - copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK; - copier_data->gtw_cfg.node_id |= SOF_IPC4_NODE_INDEX(data->dai_data); + /* + * Do not clear the node ID when this op is invoked with + * SOF_DAI_CONFIG_FLAGS_HW_FREE. It is needed to free the group_ida during + * unprepare. + */ + if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) { + copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK; + copier_data->gtw_cfg.node_id |= SOF_IPC4_NODE_INDEX(data->dai_data); + } break; case SOF_DAI_INTEL_DMIC: case SOF_DAI_INTEL_SSP: diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 760621bfc802..d7df29f2ada8 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -50,9 +50,27 @@ static int sof_widget_free_unlocked(struct snd_sof_dev *sdev, /* reset route setup status for all routes that contain this widget */ sof_reset_route_setup_status(sdev, swidget); + /* free DAI config and continue to free widget even if it fails */ + if (WIDGET_IS_DAI(swidget->id)) { + struct snd_sof_dai_config_data data; + unsigned int flags = SOF_DAI_CONFIG_FLAGS_HW_FREE; + + data.dai_data = DMA_CHAN_INVALID; + + if (tplg_ops && tplg_ops->dai_config) { + err = tplg_ops->dai_config(sdev, swidget, flags, &data); + if (err < 0) + dev_err(sdev->dev, "failed to free config for widget %s\n", + swidget->widget->name); + } + } + /* continue to disable core even if IPC fails */ - if (tplg_ops && tplg_ops->widget_free) - err = tplg_ops->widget_free(sdev, swidget); + if (tplg_ops && tplg_ops->widget_free) { + ret = tplg_ops->widget_free(sdev, swidget); + if (ret < 0 && !err) + err = ret; + } /* * disable widget core. continue to route setup status and complete flag @@ -151,8 +169,12 @@ static int sof_widget_setup_unlocked(struct snd_sof_dev *sdev, /* send config for DAI components */ if (WIDGET_IS_DAI(swidget->id)) { - unsigned int flags = SOF_DAI_CONFIG_FLAGS_NONE; + unsigned int flags = SOF_DAI_CONFIG_FLAGS_HW_PARAMS; + /* + * The config flags saved during BE DAI hw_params will be used for IPC3. IPC4 does + * not use the flags argument. + */ if (tplg_ops && tplg_ops->dai_config) { ret = tplg_ops->dai_config(sdev, swidget, flags, NULL); if (ret < 0) -- 2.39.2

2 1

[PATCH] ASoC: SOF: loader: Remove log prefixes for snd_sof_run_firmware
by Peter Ujfalusi 08 Mar '23

08 Mar '23

From: Curtis Malainey <cujomalainey(a)chromium.org> Prefixs are unneeded since log level explains the same information Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/loader.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index 81d202e5ce53..2f8555f11c03 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -123,7 +123,7 @@ int snd_sof_run_firmware(struct snd_sof_dev *sdev) "fw_version", 0444); /* errors are only due to memory allocation, not debugfs */ if (ret < 0) { - dev_err(sdev->dev, "error: snd_sof_debugfs_buf_item failed\n"); + dev_err(sdev->dev, "snd_sof_debugfs_buf_item failed\n"); return ret; } } @@ -131,7 +131,7 @@ int snd_sof_run_firmware(struct snd_sof_dev *sdev) /* perform pre fw run operations */ ret = snd_sof_dsp_pre_fw_run(sdev); if (ret < 0) { - dev_err(sdev->dev, "error: failed pre fw run op\n"); + dev_err(sdev->dev, "failed pre fw run op\n"); return ret; } -- 2.39.2

2 1

[PATCH resend] ASoC: SOF: Intel: hda-loader: use SOF helper for consistency
by Peter Ujfalusi 08 Mar '23

08 Mar '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Update code to remove mix between legacy and SOF definitions. No functionality change. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- Hi Mark, for some reason the original patch got missed: https://lore.kernel.org/alsa-devel/20230117120830.23851-1-peter.ujfalusi@li… Resending it after placing it on top of you for-next to be sure that it still applies cleanly. Regards, Peter sound/soc/sof/intel/hda-loader.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index d680562edb35..de8d9ad89bc7 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -321,13 +321,13 @@ int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream int hda_dsp_cl_boot_firmware_iccmax(struct snd_sof_dev *sdev) { struct hdac_ext_stream *iccmax_stream; - struct hdac_bus *bus = sof_to_bus(sdev); struct snd_dma_buffer dmab_bdl; int ret, ret1; u8 original_gb; /* save the original LTRP guardband value */ - original_gb = snd_hdac_chip_readb(bus, VS_LTRP) & HDA_VS_INTEL_LTRP_GB_MASK; + original_gb = snd_sof_dsp_read8(sdev, HDA_DSP_HDA_BAR, HDA_VS_INTEL_LTRP) & + HDA_VS_INTEL_LTRP_GB_MASK; /* * Prepare capture stream for ICCMAX. We do not need to store @@ -356,7 +356,8 @@ int hda_dsp_cl_boot_firmware_iccmax(struct snd_sof_dev *sdev) } /* restore the original guardband value after FW boot */ - snd_hdac_chip_updateb(bus, VS_LTRP, HDA_VS_INTEL_LTRP_GB_MASK, original_gb); + snd_sof_dsp_update8(sdev, HDA_DSP_HDA_BAR, HDA_VS_INTEL_LTRP, + HDA_VS_INTEL_LTRP_GB_MASK, original_gb); return ret; } -- 2.39.2

2 1

[PATCH] ALSA: hda: Match only Intel devices with CONTROLLER_IN_GPU()
by Bjorn Helgaas 08 Mar '23

08 Mar '23

From: Bjorn Helgaas <bhelgaas(a)google.com> CONTROLLER_IN_GPU() is clearly intended to match only Intel devices, but previously it checked only the PCI Device ID, not the Vendor ID, so it could match devices from other vendors that happened to use the same Device ID. Update CONTROLLER_IN_GPU() so it matches only Intel devices. Fixes: 535115b5ff51 ("ALSA: hda - Abort the probe without i915 binding for HSW/B") Signed-off-by: Bjorn Helgaas <bhelgaas(a)google.com> --- sound/pci/hda/hda_intel.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 81c4a45254ff..77a592f21947 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -328,14 +328,15 @@ enum { #define needs_eld_notify_link(chip) false #endif -#define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \ +#define CONTROLLER_IN_GPU(pci) (((pci)->vendor == 0x8086) && \ + (((pci)->device == 0x0a0c) || \ ((pci)->device == 0x0c0c) || \ ((pci)->device == 0x0d0c) || \ ((pci)->device == 0x160c) || \ ((pci)->device == 0x490d) || \ ((pci)->device == 0x4f90) || \ ((pci)->device == 0x4f91) || \ - ((pci)->device == 0x4f92)) + ((pci)->device == 0x4f92))) #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) -- 2.25.1

2 1

[PATCH] ALSA: hda/realtek: Fix the speaker output on Samsung Galaxy Book2 Pro
by Hamidreza H. Fard 08 Mar '23

08 Mar '23

Samsung Galaxy Book2 Pro (13" 2022 NP930XED-KA1DE) with codec SSID 144d:c868 requires the same workaround for enabling the speaker amp like other Samsung models with ALC298 code. Signed-off-by: Hamidreza H. Fard <nitocris(a)posteo.net> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3c629f4ae080..ef78553fe6f0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9539,6 +9539,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x144d, 0xc832, "Samsung Galaxy Book Flex Alpha (NP730QCJ)", ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xca03, "Samsung Galaxy Book2 Pro 360 (NP930QED)", ALC298_FIXUP_SAMSUNG_AMP), + SND_PCI_QUIRK(0x144d, 0xc868, "Samsung Galaxy Book2 Pro (NP930XED)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), base-commit: a1eeed68b1f1d23ed05f66f43058988ba734ba8f -- 2.39.2

2 1

[PATCH] ALSA: hda/realtek: fix speaker, mute/micmute LEDs not work on a HP platform
by Jeremy Szu 08 Mar '23

08 Mar '23

There is a HP platform needs ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED quirk to make mic-mute/audio-mute/speaker working. Signed-off-by: Jeremy Szu <jeremy.szu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3c629f4ae080..5d530b489c48 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9447,6 +9447,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8b8a, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b8b, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b8d, "HP", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8b8f, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b92, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8bf0, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), -- 2.38.1

2 1

[PATCH 0/2] kselftest/alsa: Log card and control information during startup
by Mark Brown 08 Mar '23

08 Mar '23

These patches help make the logs a bit more friendly to work with by adding human readable names for cards and controls alongside the numbers assigned to them even when things are working well. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Mark Brown (2): kselftest/alsa - mixer: Always log control names kselftest/alsa: Log card names during startup tools/testing/selftests/alsa/mixer-test.c | 13 +++++++++++++ tools/testing/selftests/alsa/pcm-test.c | 10 ++++++++++ 2 files changed, 23 insertions(+) --- base-commit: fe15c26ee26efa11741a7b632e9f23b01aca4cc6 change-id: 20230223-alsa-log-ctl-name-fb07f30d7217 Best regards, -- Mark Brown <broonie(a)kernel.org>

2 3

[PATCH] kselftest/alsa - mixer-test: Don't fail tests if we can't restore default
by Mark Brown 08 Mar '23

08 Mar '23

If a control has an invalid default value then we might fail to set it when restoring the default value after our write tests, for example due to correctly implemented range checks in put() operations. Currently this causes us to report the tests we were running as failed even when the operation we were trying to test is successful, making it look like there are problems where none really exist. Stop doing this, only reporting any issues during the actual test. We already have validation for the initial readback being in spec and for writing the default value back so failed tests will be reported for these controls, and we log an error on the operation that failed when we write so there will be a diagnostic warning the user that there is a problem. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- tools/testing/selftests/alsa/mixer-test.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/tools/testing/selftests/alsa/mixer-test.c b/tools/testing/selftests/alsa/mixer-test.c index 05f1749ae19d..ac5efa42d488 100644 --- a/tools/testing/selftests/alsa/mixer-test.c +++ b/tools/testing/selftests/alsa/mixer-test.c @@ -755,7 +755,6 @@ static bool test_ctl_write_valid_enumerated(struct ctl_data *ctl) static void test_ctl_write_valid(struct ctl_data *ctl) { bool pass; - int err; /* If the control is turned off let's be polite */ if (snd_ctl_elem_info_is_inactive(ctl->info)) { @@ -797,9 +796,7 @@ static void test_ctl_write_valid(struct ctl_data *ctl) } /* Restore the default value to minimise disruption */ - err = write_and_verify(ctl, ctl->def_val, NULL); - if (err < 0) - pass = false; + write_and_verify(ctl, ctl->def_val, NULL); ksft_test_result(pass, "write_valid.%d.%d\n", ctl->card->card, ctl->elem); @@ -1015,9 +1012,7 @@ static void test_ctl_write_invalid(struct ctl_data *ctl) } /* Restore the default value to minimise disruption */ - err = write_and_verify(ctl, ctl->def_val, NULL); - if (err < 0) - pass = false; + write_and_verify(ctl, ctl->def_val, NULL); ksft_test_result(pass, "write_invalid.%d.%d\n", ctl->card->card, ctl->elem); --- base-commit: fe15c26ee26efa11741a7b632e9f23b01aca4cc6 change-id: 20230224-alsa-mixer-test-restore-invalid-2a57b98aeb7f Best regards, -- Mark Brown <broonie(a)kernel.org>

2 1

[PATCH] ALSA: hda: intel-dsp-config: add MTL PCI id
by Bard Liao 08 Mar '23

08 Mar '23

Use SOF as default audio driver. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Gongjun Song <gongjun.song(a)intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/hda/intel-dsp-config.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index ae31bb127594..317bdf6dcbef 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -472,6 +472,15 @@ static const struct config_entry config_table[] = { }, #endif +/* Meteor Lake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_METEORLAKE) + /* Meteorlake-P */ + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, + .device = 0x7e28, + }, +#endif + }; static const struct config_entry *snd_intel_dsp_find_config -- 2.25.1

2 1

[PATCH 0/130+] ALSA: emu10k1 & more
by Oswald Buddenhagen 08 Mar '23

08 Mar '23

moin, i got myself an old e-mu 0404 card, because i found the analog hardware well hackable for dc operation. then i found the driver to be lacking, and got hacking that, too. hence this blast from the past. this is a huge branch with many internal dependencies, so i'm assuming i'm not supposed to post a patch series. instead it's on https://github.com/ossilator/linux/tree/ossis-alsa - let me know if another review platform is preferable. apart from the actual driver work, this contains some alsa patches unrelated to emu10k1, some patches which really shouldn't be merged via the alsa route (if at all), and some patches which are just collections of questions (to which i hope to receive answers on github). this is of course meant to be cleaned up in due time. feel free to cherry-pick the patches you deem ready; i'll happily rebase the branch. some of these patches are not (fully) tested due to lack of hardware. in particular, someone should check that SB Live! and Audigy cards still work. then there is a whole bunch of e-mu card specific functionality which targets the "bigger members of the family". all patches that need testing are clearly marked as such. if someone wants to provide me with hardware (temporarily or permanently), go ahead (i'm in berlin, germany). on CC is everyone who is recorded to have hacked the emu1010 part of the driver or tested something. i fully expect half the addresses to bounce, and the remainder to wonder what i'm smoking. ;-) when you reply, please do so only to the list and me. regards

2 1

[PATCH v3] ASoC: tas2781: Add tas2781 driver
by Kevin Lu 08 Mar '23

08 Mar '23

Create tas2781 driver. --- Changes in v3: - fixed issues reported-by: kernel test robot <lkp(a)intel.com> - correct all the inconsistent indenting warnings - correct uninitialized symbol 'pwr_state' - correct unsigned 'val' is never less than zero - correct 'tas_dev' could be null Signed-off-by: Kevin Lu <luminlong(a)139.com> --- sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2781-dsp.c | 2261 ++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2781-dsp.h | 178 +++ sound/soc/codecs/tas2781-i2c.c | 1913 +++++++++++++++++++++++++++ sound/soc/codecs/tas2781.h | 176 +++ 6 files changed, 4542 insertions(+) create mode 100644 sound/soc/codecs/tas2781-dsp.c create mode 100644 sound/soc/codecs/tas2781-dsp.h create mode 100644 sound/soc/codecs/tas2781-i2c.c create mode 100644 sound/soc/codecs/tas2781.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0f9d714..9016c48 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -223,6 +223,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2764 imply SND_SOC_TAS2770 imply SND_SOC_TAS2780 + imply SND_SOC_TAS2781 imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -1579,6 +1580,17 @@ config SND_SOC_TAS2780 Enable support for Texas Instruments TAS2780 high-efficiency digital input mono Class-D audio power amplifiers. +config SND_SOC_TAS2781 + tristate "Texas Instruments TAS2781 speaker amplifier" + depends on I2C + select REGMAP_I2C + select CRC8 + help + Enable support for Texas Instruments TAS2781 Smart Amplifier + Digital input mono Class-D and DSP-inside audio power amplifiers. + Note the TAS2781 driver implements a flexible and configurable + algo coff setting, for one, two, even multiple TAS2781 chips. + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 71d3ce5..49f912a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -355,6 +355,7 @@ snd-soc-tas2552-objs := tas2552.o snd-soc-tas2562-objs := tas2562.o snd-soc-tas2764-objs := tas2764.o snd-soc-tas2780-objs := tas2780.o +snd-soc-tas2781-objs := tas2781-i2c.o tas2781-dsp.o # Mux snd-soc-simple-mux-objs := simple-mux.o @@ -606,6 +607,7 @@ obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o obj-$(CONFIG_SND_SOC_TAS2562) += snd-soc-tas2562.o obj-$(CONFIG_SND_SOC_TAS2764) += snd-soc-tas2764.o obj-$(CONFIG_SND_SOC_TAS2780) += snd-soc-tas2780.o +obj-$(CONFIG_SND_SOC_TAS2781) += snd-soc-tas2781.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2781-dsp.c b/sound/soc/codecs/tas2781-dsp.c new file mode 100644 index 0000000..0429021 --- /dev/null +++ b/sound/soc/codecs/tas2781-dsp.c @@ -0,0 +1,2261 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier + * + * Copyright (C) 2022 - 2023 Texas Instruments Incorporated + * https://www.ti.com + * + * The TAS2781 driver implements a flexible and configurable algo coff setting + * for one, two, even multiple TAS2781 chips. + * + * Author: Shenghao Ding <shenghao-ding(a)ti.com> + * Kevin Lu <kevin-lu(a)ti.com> + */ + +#include <linux/regmap.h> +#include <linux/i2c.h> +#include <linux/string.h> +#include <linux/crc8.h> +#include <linux/slab.h> +#include <linux/fs.h> +#include <linux/firmware.h> + +#include "tas2781.h" + +#define ERROR_PRAM_CRCCHK 0x0000000 +#define ERROR_YRAM_CRCCHK 0x0000001 +#define BINFILE_VER 0 +#define DRV_VER 1 +#define PPC_DRIVER_CRCCHK 0x00000200 + +#define TAS2781_SA_COEFF_SWAP_REG TASDEVICE_REG(0, 0x35, 0x2c) +#define TAS2781_YRAM_BOOK1 140 +#define TAS2781_YRAM1_PAGE 42 +#define TAS2781_YRAM1_START_REG 88 + +#define TAS2781_YRAM2_START_PAGE 43 +#define TAS2781_YRAM2_END_PAGE 49 +#define TAS2781_YRAM2_START_REG 8 +#define TAS2781_YRAM2_END_REG 127 + +#define TAS2781_YRAM3_PAGE 50 +#define TAS2781_YRAM3_START_REG 8 +#define TAS2781_YRAM3_END_REG 27 + +/*should not include B0_P53_R44-R47 */ +#define TAS2781_YRAM_BOOK2 0 +#define TAS2781_YRAM4_START_PAGE 50 +#define TAS2781_YRAM4_END_PAGE 60 + +#define TAS2781_YRAM5_PAGE 61 +#define TAS2781_YRAM5_START_REG 8 +#define TAS2781_YRAM5_END_REG 27 + +#define TASDEVICE_MAXPROGRAM_NUM_KERNEL 5 +#define TASDEVICE_MAXCONFIG_NUM_KERNEL_MULTIPLE_AMPS 64 +#define TASDEVICE_MAXCONFIG_NUM_KERNEL 10 +#define MAIN_ALL_DEVICES_1X 0x01 +#define MAIN_DEVICE_A_1X 0x02 +#define MAIN_DEVICE_B_1X 0x03 +#define MAIN_DEVICE_C_1X 0x04 +#define MAIN_DEVICE_D_1X 0x05 +#define COEFF_DEVICE_A_1X 0x12 +#define COEFF_DEVICE_B_1X 0x13 +#define COEFF_DEVICE_C_1X 0x14 +#define COEFF_DEVICE_D_1X 0x15 +#define PRE_DEVICE_A_1X 0x22 +#define PRE_DEVICE_B_1X 0x23 +#define PRE_DEVICE_C_1X 0x24 +#define PRE_DEVICE_D_1X 0x25 + +struct tas_crc { + unsigned char offset; + unsigned char len; +}; + +const char deviceNumber[TASDEVICE_DSP_TAS_MAX_DEVICE] = { + 1, 2, 1, 2, 1, 1, 0, 2, 4, 3, 1, 2, 3, 4 +}; + +static int fw_parse_block_data_kernel(struct tasdevice_fw *tas_fmw, + struct tasdev_blk *block, const struct firmware *fmw, int offset) +{ + const unsigned char *data = fmw->data; + + if (offset + 4 > fmw->size) { + dev_err(tas_fmw->dev, "%s: File Size error\n", __func__); + offset = -1; + goto out; + } + block->type = SMS_HTONL(data[offset], + data[offset + 1], data[offset + 2], + data[offset + 3]); + offset += 4; + + if (offset + 1 > fmw->size) { + dev_err(tas_fmw->dev, "%s: mbPChkSumPresent error\n", + __func__); + offset = -1; + goto out; + } + block->is_pchksum_present = data[offset]; + offset++; + + if (offset + 1 > fmw->size) { + dev_err(tas_fmw->dev, "%s: mnPChkSum error\n", __func__); + offset = -1; + goto out; + } + block->pchksum = data[offset]; + offset++; + + if (offset + 1 > fmw->size) { + dev_err(tas_fmw->dev, "%s: mbYChkSumPresent error\n", + __func__); + offset = -1; + goto out; + } + block->is_ychksum_present = data[offset]; + offset++; + + if (offset + 1 > fmw->size) { + dev_err(tas_fmw->dev, "%s: mnYChkSum error\n", __func__); + offset = -1; + goto out; + } + block->ychksum = data[offset]; + offset++; + + if (offset + 4 > fmw->size) { + dev_err(tas_fmw->dev, "%s: blk_size error\n", __func__); + offset = -1; + goto out; + } + block->blk_size = SMS_HTONL(data[offset], data[offset + 1], + data[offset + 2], data[offset + 3]); + offset += 4; + + if (offset + 4 > fmw->size) { + dev_err(tas_fmw->dev, "%s: nSublocks error\n", __func__); + offset = -1; + goto out; + } + block->n_subblocks = SMS_HTONL(data[offset], data[offset + 1], + data[offset + 2], data[offset + 3]); + offset += 4; + + block->data = kzalloc(block->blk_size, GFP_KERNEL); + if (block->data == NULL) { + offset = -1; + goto out; + } + memcpy(block->data, &data[offset], block->blk_size); + offset += block->blk_size; +out: + return offset; +} + +static int fw_parse_data_kernel(struct tasdevice_fw *tas_fmw, + struct tasdevice_data *img_data, const struct firmware *fmw, + int offset) +{ + const unsigned char *data = fmw->data; + unsigned int n_block; + + if (offset + 4 > fmw->size) { + dev_err(tas_fmw->dev, "%s: File Size error\n", __func__); + offset = -1; + goto out; + } + img_data->n_blk = SMS_HTONL(data[offset], + data[offset + 1], data[offset + 2], data[offset + 3]); + offset += 4; + + img_data->dev_blks = + kcalloc(img_data->n_blk, sizeof(struct tasdev_blk), + GFP_KERNEL); + if (img_data->dev_blks == NULL) { + dev_err(tas_fmw->dev, "%s: FW memory failed!\n", __func__); + goto out; + } + + for (n_block = 0; n_block < img_data->n_blk; n_block++) { + offset = fw_parse_block_data_kernel(tas_fmw, + &(img_data->dev_blks[n_block]), fmw, offset); + if (offset < 0) { + offset = -1; + goto out; + } + } +out: + return offset; +} + +static int fw_parse_program_data_kernel( + struct tasdevice_priv *tas_dev, struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset) +{ + struct tasdevice_prog *program; + unsigned int n_program = 0; + + for (n_program = 0; n_program < tas_fmw->n_programs; n_program++) { + program = &(tas_fmw->programs[n_program]); + if (offset + 64 > fmw->size) { + dev_err(tas_dev->dev, "%s: mpName error\n", + __func__); + offset = -1; + goto out; + } + offset += 64; + + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnAppMode error\n", + __func__); + offset = -1; + goto out; + } + offset++; + + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnPDMI2SMode error\n", + __func__); + offset = -1; + goto out; + } + offset++; + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnISnsPD error\n", + __func__); + offset = -1; + goto out; + } + offset++; + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnVSnsPD error\n", + __func__); + offset = -1; + goto out; + } + offset++; + //skip 3-byte reserved + offset += 3; + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnPowerLDG error\n", + __func__); + offset = -1; + goto out; + } + offset++; + + offset = fw_parse_data_kernel(tas_fmw, &(program->dev_data), + fmw, offset); + if (offset < 0) + goto out; + } +out: + return offset; +} + +static int fw_parse_configuration_data_kernel( + struct tasdevice_priv *tas_dev, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + const unsigned char *data = fmw->data; + unsigned int n_configs; + struct tasdevice_config *config; + + for (n_configs = 0; n_configs < tas_fmw->n_configurations; + n_configs++) { + config = + &(tas_fmw->configs[n_configs]); + if (offset + 64 > fmw->size) { + dev_err(tas_dev->dev, "%s: mpName error\n", + __func__); + offset = -1; + goto out; + } + memcpy(config->name, &data[offset], 64); + offset += 64; + + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, "%s: orientation error\n", + __func__); + offset = -1; + goto out; + } + offset++; + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnDevices error\n", + __func__); + offset = -1; + goto out; + } + offset += 1; + + if (offset + 2 > fmw->size) { + dev_err(tas_dev->dev, "%s: mProgram error\n", + __func__); + offset = -1; + goto out; + } + offset += 2; + + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnSamplingRate error\n", + __func__); + offset = -1; + goto out; + } + offset += 4; + + if (offset + 2 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnPLLSrc error\n", + __func__); + offset = -1; + goto out; + } + offset += 2; + + if (offset + 2 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnFsRate error\n", + __func__); + offset = -1; + goto out; + } + offset += 2; + + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnPLLSrcRate error\n", + __func__); + offset = -1; + goto out; + } + offset += 4; + offset = fw_parse_data_kernel(tas_fmw, + &(config->dev_data), fmw, offset); + if (offset < 0) + goto out; + } +out: + return offset; +} + +static int fw_parse_variable_header_kernel( + struct tasdevice_priv *tas_dev, const struct firmware *fmw, int offset) +{ + struct tasdevice_fw *tas_fmw = tas_dev->fmw; + struct tasdevice_dspfw_hdr *fw_hdr = &(tas_fmw->fw_hdr); + const unsigned char *buf = fmw->data; + struct tasdevice_prog *program; + struct tasdevice_config *config; + unsigned int n_program = 0, n_configs = 0; + unsigned short max_confs = TASDEVICE_MAXCONFIG_NUM_KERNEL; + + if (offset + 2 > fmw->size) { + dev_err(tas_dev->dev, "%s: File Size error\n", __func__); + offset = -1; + goto out; + } + fw_hdr->device_family = SMS_HTONS(buf[offset], buf[offset + 1]); + if (fw_hdr->device_family != 0) { + dev_err(tas_dev->dev, "ERROR:%s:not TAS device\n", __func__); + offset = -1; + goto out; + } + offset += 2; + if (offset + 2 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnDevice error\n", __func__); + offset = -1; + goto out; + } + fw_hdr->device = SMS_HTONS(buf[offset], buf[offset + 1]); + if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || + fw_hdr->device == 6) { + dev_err(tas_dev->dev, "ERROR:%s: not support device %d\n", + __func__, fw_hdr->device); + offset = -1; + goto out; + } + offset += 2; + fw_hdr->ndev = deviceNumber[fw_hdr->device]; + + if (fw_hdr->ndev != tas_dev->ndev) { + dev_err(tas_dev->dev, + "%s: ndev(%u) in dspbin dismatch ndev(%u) in DTS\n", + __func__, fw_hdr->ndev, tas_dev->ndev); + offset = -1; + goto out; + } + + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnPrograms error\n", __func__); + offset = -1; + goto out; + } + tas_fmw->n_programs = SMS_HTONL(buf[offset], buf[offset + 1], + buf[offset + 2], buf[offset + 3]); + offset += 4; + + if (tas_fmw->n_programs == 0 || tas_fmw->n_programs > + TASDEVICE_MAXPROGRAM_NUM_KERNEL) { + dev_err(tas_dev->dev, "%s: mnPrograms is invalid\n", __func__); + offset = -1; + goto out; + } + + if (offset + 4 * TASDEVICE_MAXPROGRAM_NUM_KERNEL > fmw->size) { + dev_err(tas_dev->dev, "%s: mpPrograms error\n", __func__); + offset = -1; + goto out; + } + + tas_fmw->programs = + kcalloc(tas_fmw->n_programs, + sizeof(struct tasdevice_prog), GFP_KERNEL); + if (tas_fmw->programs == NULL) { + dev_err(tas_dev->dev, "%s: mpPrograms memory failed!\n", + __func__); + offset = -1; + goto out; + } + + for (n_program = 0; n_program < tas_fmw->n_programs; n_program++) { + program = &(tas_fmw->programs[n_program]); + program->prog_size = SMS_HTONL(buf[offset], buf[offset + 1], + buf[offset + 2], buf[offset + 3]); + offset += 4; + } + offset += (4 * (5 - n_program)); + + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnConfigurations error\n", + __func__); + offset = -1; + goto out; + } + tas_fmw->n_configurations = SMS_HTONL(buf[offset], buf[offset + 1], + buf[offset + 2], buf[offset + 3]); + offset += 4; + max_confs = (fw_hdr->ndev >= 4) ? + TASDEVICE_MAXCONFIG_NUM_KERNEL_MULTIPLE_AMPS : + TASDEVICE_MAXCONFIG_NUM_KERNEL; + if (tas_fmw->n_configurations == 0 || + tas_fmw->n_configurations > max_confs) { + dev_err(tas_dev->dev, "%s: mnConfigurations is invalid\n", + __func__); + offset = -1; + goto out; + } + + if (offset + 4 * max_confs > fmw->size) { + dev_err(tas_dev->dev, "%s: mpConfigurations error\n", + __func__); + offset = -1; + goto out; + } + + tas_fmw->configs = kcalloc(tas_fmw->n_configurations, + sizeof(struct tasdevice_config), GFP_KERNEL); + if (tas_fmw->configs == NULL) { + offset = -1; + goto out; + } + + for (n_configs = 0; n_configs < tas_fmw->n_programs; + n_configs++) { + config = + &(tas_fmw->configs[n_configs]); + config->cfg_size = SMS_HTONL(buf[offset], + buf[offset + 1], buf[offset + 2], buf[offset + 3]); + offset += 4; + } + + offset += (4 * (max_confs - n_configs)); +out: + return offset; +} + +static int tasdevice_load_block_kernel( + struct tasdevice_priv *tasdevice, struct tasdev_blk *block) +{ + int ret = 0; + + unsigned char *data = block->data; + unsigned int i = 0, length = 0; + const unsigned int blk_size = block->blk_size; + unsigned char dev_idx = 0; + struct tasdevice_dspfw_hdr *fw_hdr = &(tasdevice->fmw->fw_hdr); + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr = &(fw_hdr->fixed_hdr); + + if (fw_fixed_hdr->ppcver >= PPC3_VERSION) { + switch (block->type) { + case MAIN_ALL_DEVICES_1X: + dev_idx = 0|0x80; + break; + case MAIN_DEVICE_A_1X: + dev_idx = 1|0x80; + break; + case COEFF_DEVICE_A_1X: + case PRE_DEVICE_A_1X: + dev_idx = 1|0xC0; + break; + case MAIN_DEVICE_B_1X: + dev_idx = 2|0x80; + break; + case COEFF_DEVICE_B_1X: + case PRE_DEVICE_B_1X: + dev_idx = 2|0xC0; + break; + case MAIN_DEVICE_C_1X: + dev_idx = 3|0x80; + break; + case COEFF_DEVICE_C_1X: + case PRE_DEVICE_C_1X: + dev_idx = 3|0xC0; + break; + case MAIN_DEVICE_D_1X: + dev_idx = 4|0x80; + break; + case COEFF_DEVICE_D_1X: + case PRE_DEVICE_D_1X: + dev_idx = 4|0xC0; + break; + default: + dev_info(tasdevice->dev, + "%s: load block: Other Type = 0x%02x\n", + __func__, block->type); + break; + } + } else { + switch (block->type) { + case MAIN_ALL_DEVICES: + dev_idx = 0|0x80; + break; + case MAIN_DEVICE_A: + dev_idx = 1|0x80; + break; + case COEFF_DEVICE_A: + case PRE_DEVICE_A: + dev_idx = 1|0xC0; + break; + case MAIN_DEVICE_B: + dev_idx = 2|0x80; + break; + case COEFF_DEVICE_B: + case PRE_DEVICE_B: + dev_idx = 2|0xC0; + break; + case MAIN_DEVICE_C: + dev_idx = 3|0x80; + break; + case COEFF_DEVICE_C: + case PRE_DEVICE_C: + dev_idx = 3|0xC0; + break; + case MAIN_DEVICE_D: + dev_idx = 4|0x80; + break; + case COEFF_DEVICE_D: + case PRE_DEVICE_D: + dev_idx = 4|0xC0; + break; + default: + dev_info(tasdevice->dev, + "%s: load block: Other Type = 0x%02x\n", + __func__, block->type); + break; + } + } + + for (i = 0; i < block->n_subblocks; i++) { + int rc = tasdevice_process_block(tasdevice, data + length, + dev_idx, blk_size - length); + if (rc < 0) { + dev_err(tasdevice->dev, + "%s: ERROR:%u %u sublock write error\n", + __func__, length, blk_size); + break; + } + length += (unsigned int)rc; + if (blk_size < length) { + dev_err(tasdevice->dev, + "%s: ERROR:%u %u out of memory\n", + __func__, length, blk_size); + break; + } + } + + return ret; +} + +static int fw_parse_variable_header_git(struct tasdevice_priv *tas_dev, + const struct firmware *fmw, int offset) +{ + const unsigned char *buf = fmw->data; + struct tasdevice_fw *tas_fmw = tas_dev->fmw; + struct tasdevice_dspfw_hdr *fw_hdr = &(tas_fmw->fw_hdr); + int i = strlen((char *)&buf[offset]); + + i++; + + if (offset + i > fmw->size) { + dev_err(tas_dev->dev, "%s: File Size error\n", __func__); + offset = -1; + goto out; + } + + offset += i; + + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: File Size error\n", __func__); + offset = -1; + goto out; + } + fw_hdr->device_family = SMS_HTONL(buf[offset], buf[offset + 1], + buf[offset + 2], buf[offset + 3]); + if (fw_hdr->device_family != 0) { + dev_err(tas_dev->dev, "ERROR:%s: not TAS device\n", __func__); + offset = -1; + goto out; + } + offset += 4; + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: File Size error\n", __func__); + offset = -1; + goto out; + } + fw_hdr->device = SMS_HTONL(buf[offset], buf[offset + 1], + buf[offset + 2], buf[offset + 3]); + if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || + fw_hdr->device == 6) { + dev_err(tas_dev->dev, "ERROR:%s: not support device %d\n", + __func__, fw_hdr->device); + offset = -1; + goto out; + } + offset += 4; + fw_hdr->ndev = deviceNumber[fw_hdr->device]; + if (fw_hdr->ndev != tas_dev->ndev) { + dev_err(tas_dev->dev, + "%s: ndev(%u) in dspbin dismatch ndev(%u) in DTS\n", + __func__, fw_hdr->ndev, + tas_dev->ndev); + offset = -1; + } + +out: + return offset; +} + +static int fw_parse_variable_hdr_cal(struct tasdevice_priv *tas_dev, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + const unsigned char *buf = fmw->data; + struct tasdevice_dspfw_hdr *fw_hdr = &(tas_fmw->fw_hdr); + int i = strlen((char *)&buf[offset]); + + i++; + + if (offset + i > fmw->size) { + dev_err(tas_dev->dev, "%s: File Size error\n", __func__); + offset = -1; + goto out; + } + + offset += i; + + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnDeviceFamily error\n", __func__); + offset = -1; + goto out; + } + fw_hdr->device_family = SMS_HTONL(buf[offset], buf[offset + 1], + buf[offset + 2], buf[offset + 3]); + if (fw_hdr->device_family != 0) { + dev_err(tas_dev->dev, "ERROR:%s: not TAS device\n", __func__); + offset = -1; + goto out; + } + offset += 4; + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnDevice error\n", __func__); + offset = -1; + goto out; + } + fw_hdr->device = SMS_HTONL(buf[offset], buf[offset + 1], + buf[offset + 2], buf[offset + 3]); + if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || + fw_hdr->device == 6) { + dev_err(tas_dev->dev, "ERROR:%s: not support device %d\n", + __func__, fw_hdr->device); + offset = -1; + goto out; + } + offset += 4; + fw_hdr->ndev = deviceNumber[fw_hdr->device]; + if (fw_hdr->ndev != 1) { + dev_err(tas_dev->dev, + "%s: calbin must be 1, but currently ndev(%u)\n", + __func__, fw_hdr->ndev); + offset = -1; + } + +out: + return offset; +} + +static inline void tas2781_clear_calfirmware(struct tasdevice_fw + *tas_fmw) +{ + int i = 0; + unsigned int n_blk = 0; + + if (tas_fmw->calibrations) { + struct tasdevice_calibration *calibration; + + for (i = 0; i < tas_fmw->n_calibrations; i++) { + calibration = &(tas_fmw->calibrations[i]); + if (calibration) { + struct tasdevice_data *im = + &(calibration->dev_data); + + if (im->dev_blks) { + struct tasdev_blk *block; + + for (n_blk = 0; n_blk < im->n_blk; + n_blk++) { + block = &(im->dev_blks[n_blk]); + kfree(block->data); + } + kfree(im->dev_blks); + } + } + } + kfree(tas_fmw->calibrations); + } + kfree(tas_fmw); +} + +static int fw_parse_block_data(struct tasdevice_fw *tas_fmw, + struct tasdev_blk *block, const struct firmware *fmw, int offset) +{ + unsigned char *data = (unsigned char *)fmw->data; + int n; + + if (offset + 4 > fmw->size) { + dev_err(tas_fmw->dev, "%s: mnType error\n", __func__); + offset = -1; + goto out; + } + block->type = SMS_HTONL(data[offset], data[offset + 1], + data[offset + 2], data[offset + 3]); + offset += 4; + + if (tas_fmw->fw_hdr.fixed_hdr.drv_ver >= + PPC_DRIVER_CRCCHK) { + if (offset + 1 > fmw->size) { + dev_err(tas_fmw->dev, "%s: mbPChkSumPresent error\n", + __func__); + offset = -1; + goto out; + } + block->is_pchksum_present = data[offset]; + offset++; + + if (offset + 1 > fmw->size) { + dev_err(tas_fmw->dev, "%s: mnPChkSum error\n", + __func__); + offset = -1; + goto out; + } + block->pchksum = data[offset]; + offset++; + + if (offset + 1 > fmw->size) { + dev_err(tas_fmw->dev, "%s: mbYChkSumPresent error\n", + __func__); + offset = -1; + goto out; + } + block->is_ychksum_present = data[offset]; + offset++; + + if (offset + 1 > fmw->size) { + dev_err(tas_fmw->dev, "%s: mnYChkSum error\n", + __func__); + offset = -1; + goto out; + } + block->ychksum = data[offset]; + offset++; + } else { + block->is_pchksum_present = 0; + block->is_ychksum_present = 0; + } + if (offset + 4 > fmw->size) { + dev_err(tas_fmw->dev, "%s: mnCommands error\n", __func__); + offset = -1; + goto out; + } + block->n_cmds = SMS_HTONL(data[offset], data[offset + 1], + data[offset + 2], data[offset + 3]); + offset += 4; + + n = block->n_cmds * 4; + if (offset + n > fmw->size) { + dev_err(tas_fmw->dev, + "%s: File Size(%lu) error offset = %d n = %d\n", + __func__, (unsigned long)fmw->size, offset, n); + offset = -1; + goto out; + } + block->data = kmemdup(&data[offset], n, GFP_KERNEL); + if (block->data == NULL) { + offset = -1; + goto out; + } + offset += n; +out: + return offset; +} + +static int fw_parse_data(struct tasdevice_fw *tas_fmw, + struct tasdevice_data *img_data, const struct firmware *fmw, + int offset) +{ + const unsigned char *data = (unsigned char *)fmw->data; + int n = 0; + unsigned int n_block; + + if (offset + 64 > fmw->size) { + dev_err(tas_fmw->dev, "%s: mpName error\n", __func__); + n = -1; + goto out; + } + memcpy(img_data->name, &data[offset], 64); + offset += 64; + + n = strlen((char *)&data[offset]); + n++; + if (offset + n > fmw->size) { + dev_err(tas_fmw->dev, "%s: mpDescription error\n", __func__); + offset = -1; + goto out; + } + offset += n; + + if (offset + 2 > fmw->size) { + dev_err(tas_fmw->dev, "%s: mnBlocks error\n", __func__); + offset = -1; + goto out; + } + img_data->n_blk = SMS_HTONS(data[offset], data[offset + 1]); + offset += 2; + + img_data->dev_blks = + kcalloc(img_data->n_blk, sizeof(struct tasdev_blk), + GFP_KERNEL); + if (img_data->dev_blks == NULL) { + dev_err(tas_fmw->dev, "%s: FW memory failed!\n", __func__); + goto out; + } + for (n_block = 0; n_block < img_data->n_blk; n_block++) { + offset = fw_parse_block_data(tas_fmw, + &(img_data->dev_blks[n_block]), fmw, offset); + if (offset < 0) { + offset = -1; + goto out; + } + } +out: + return offset; +} + +static int fw_parse_calibration_data(struct tasdevice_priv *tas_dev, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + unsigned char *data = (unsigned char *)fmw->data; + unsigned int n = 0; + unsigned int n_calibration = 0; + struct tasdevice_calibration *calibration = NULL; + + if (offset + 2 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnCalibrations error\n", __func__); + offset = -1; + goto out; + } + tas_fmw->n_calibrations = SMS_HTONS(data[offset], data[offset + 1]); + offset += 2; + + if (tas_fmw->n_calibrations != 1) { + dev_err(tas_dev->dev, + "%s: only support one calibraiton(%d)!\n", + __func__, tas_fmw->n_calibrations); + goto out; + } + + tas_fmw->calibrations = + kcalloc(tas_fmw->n_calibrations, + sizeof(struct tasdevice_calibration), GFP_KERNEL); + if (tas_fmw->calibrations == NULL) { + dev_err(tas_dev->dev, "%s: mpCalibrations memory failed!\n", + __func__); + offset = -1; + goto out; + } + for (n_calibration = 0; n_calibration < tas_fmw->n_calibrations; + n_calibration++) { + if (offset + 64 > fmw->size) { + dev_err(tas_dev->dev, "%s: mpCalibrations error\n", + __func__); + offset = -1; + goto out; + } + calibration = &(tas_fmw->calibrations[n_calibration]); + offset += 64; + + n = strlen((char *)&data[offset]); + n++; + if (offset + n > fmw->size) { + dev_err(tas_dev->dev, "%s: mpDescription error\n", + __func__); + offset = -1; + goto out; + } + offset += n; + + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, + "%s: mnProgram error, offset = %d\n", __func__, + offset); + offset = -1; + goto out; + } + offset++; + + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, + "%s: mnConfiguration error, offset = %d\n", + __func__, + offset); + offset = -1; + goto out; + } + offset++; + + offset = fw_parse_data(tas_fmw, &(calibration->dev_data), fmw, + offset); + if (offset < 0) + goto out; + } + +out: + + return offset; +} + +static int fw_parse_program_data(struct tasdevice_priv *tas_dev, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + struct tasdevice_prog *program; + unsigned char *buf = (unsigned char *)fmw->data; + int n_program = 0; + + if (offset + 2 > fmw->size) { + dev_err(tas_dev->dev, "%s: File Size error\n", __func__); + offset = -1; + goto out; + } + tas_fmw->n_programs = SMS_HTONS(buf[offset], buf[offset + 1]); + offset += 2; + + if (tas_fmw->n_programs == 0) { + dev_err(tas_dev->dev, "%s: mnPrograms is null, maybe calbin\n", + __func__); + //Do not "offset = -1;", because of calbin + goto out; + } + + tas_fmw->programs = + kcalloc(tas_fmw->n_programs, sizeof(struct tasdevice_prog), + GFP_KERNEL); + if (tas_fmw->programs == NULL) { + offset = -1; + goto out; + } + for (n_program = 0; n_program < tas_fmw->n_programs; n_program++) { + int n = 0; + + program = &(tas_fmw->programs[n_program]); + if (offset + 64 > fmw->size) { + dev_err(tas_dev->dev, "%s: mpName error\n", __func__); + offset = -1; + goto out; + } + offset += 64; + + n = strlen((char *)&buf[offset]); + n++; + if (offset + n > fmw->size) { + dev_err(tas_dev->dev, "%s: mpDescription error\n", + __func__); + offset = -1; + goto out; + } + + offset += n; + + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnAppMode error\n", + __func__); + offset = -1; + goto out; + } + offset++; + + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnPDMI2SMode error\n", + __func__); + offset = -1; + goto out; + } + offset++; + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnISnsPD error\n", + __func__); + offset = -1; + goto out; + } + offset++; + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnVSnsPD error\n", + __func__); + offset = -1; + goto out; + } + offset++; + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnPowerLDG error\n", + __func__); + offset = -1; + goto out; + } + offset++; + + offset = fw_parse_data(tas_fmw, &(program->dev_data), fmw, + offset); + if (offset < 0) + goto out; + } +out: + return offset; +} + +static int fw_parse_configuration_data( + struct tasdevice_priv *tas_dev, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset) +{ + unsigned char *data = (unsigned char *)fmw->data; + int n; + unsigned int n_configs; + struct tasdevice_config *config; + + if (offset + 2 > fmw->size) { + dev_err(tas_dev->dev, "%s: File Size error\n", __func__); + offset = -1; + goto out; + } + tas_fmw->n_configurations = SMS_HTONS(data[offset], data[offset + 1]); + offset += 2; + + if (tas_fmw->n_configurations == 0) { + dev_err(tas_dev->dev, "%s: mnConfigurations is zero\n", + __func__); + //Do not "offset = -1;", because of calbin + goto out; + } + tas_fmw->configs = + kcalloc(tas_fmw->n_configurations, + sizeof(struct tasdevice_config), GFP_KERNEL); + + for (n_configs = 0; n_configs < tas_fmw->n_configurations; + n_configs++) { + config = + &(tas_fmw->configs[n_configs]); + if (offset + 64 > fmw->size) { + dev_err(tas_dev->dev, "%s: File Size error\n", + __func__); + offset = -1; + goto out; + } + memcpy(config->name, &data[offset], 64); + offset += 64; + + n = strlen((char *)&data[offset]); + n++; + if (offset + n > fmw->size) { + dev_err(tas_dev->dev, "%s: mpDescription error\n", + __func__); + offset = -1; + goto out; + } + + offset += n; + if (offset + 2 > fmw->size) { + dev_err(tas_dev->dev, + "%s: mnDevice_orientation error\n", + __func__); + offset = -1; + goto out; + } + offset += 2; + + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, "%s: mProgram error\n", + __func__); + offset = -1; + goto out; + } + offset++; + + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnSamplingRate error\n", + __func__); + offset = -1; + goto out; + } + offset += 4; + + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnPLLSrc error\n", + __func__); + offset = -1; + goto out; + } + offset++; + + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnPLLSrcRate error\n", + __func__); + offset = -1; + goto out; + } + offset += 4; + + if (offset + 2 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnFsRate error\n", + __func__); + offset = -1; + goto out; + } + offset += 2; + + offset = fw_parse_data(tas_fmw, &(config->dev_data), + fmw, offset); + if (offset < 0) + goto out; + } +out: + return offset; +} + +static int fw_parse_header(struct tasdevice_priv *tas_dev, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + struct tasdevice_dspfw_hdr *fw_hdr = &(tas_fmw->fw_hdr); + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr = &(fw_hdr->fixed_hdr); + const unsigned char *buf = (unsigned char *)fmw->data; + const unsigned char magic_number[] = { 0x35, 0x35, 0x35, 0x32 }; + + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: File Size error\n", __func__); + offset = -EINVAL; + goto out; + } + if (memcmp(&buf[offset], magic_number, 4)) { + dev_err(tas_dev->dev, "%s: Magic number doesn't match", + __func__); + offset = -EINVAL; + goto out; + } + offset += 4; + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnFWSize error\n", __func__); + offset = -1; + goto out; + } + fw_fixed_hdr->fwsize = SMS_HTONL(buf[offset], + buf[offset + 1], buf[offset + 2], buf[offset + 3]); + offset += 4; + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: File Size error\n", __func__); + offset = -1; + goto out; + } + if (fw_fixed_hdr->fwsize != fmw->size) { + dev_err(tas_dev->dev, "File size not match, %lu %u", + (unsigned long)fmw->size, fw_fixed_hdr->fwsize); + offset = -1; + goto out; + } + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnChecksum error\n", __func__); + offset = -1; + goto out; + } + offset += 4; + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnPPCVersion error\n", __func__); + offset = -1; + goto out; + } + fw_fixed_hdr->ppcver = SMS_HTONL(buf[offset], + buf[offset + 1], buf[offset + 2], buf[offset + 3]); + offset += 4; + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnFWVersion error\n", __func__); + offset = -1; + goto out; + } + offset += 4; + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnDriverVersion error\n", __func__); + offset = -1; + goto out; + } + fw_fixed_hdr->drv_ver = SMS_HTONL(buf[offset], + buf[offset + 1], buf[offset + 2], buf[offset + 3]); + offset += 4; + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnTimeStamp error\n", __func__); + offset = -1; + goto out; + } + offset += 4; + if (offset + 64 > fmw->size) { + dev_err(tas_dev->dev, "%s: mpDDCName error\n", __func__); + offset = -1; + goto out; + } + offset += 64; + + out: + return offset; +} + +static const unsigned char crc8_lookup_table[CRC8_TABLE_SIZE] = { + 0x00, 0x4D, 0x9A, 0xD7, 0x79, 0x34, 0xE3, 0xAE, + 0xF2, 0xBF, 0x68, 0x25, 0x8B, 0xC6, 0x11, 0x5C, + 0xA9, 0xE4, 0x33, 0x7E, 0xD0, 0x9D, 0x4A, 0x07, + 0x5B, 0x16, 0xC1, 0x8C, 0x22, 0x6F, 0xB8, 0xF5, + 0x1F, 0x52, 0x85, 0xC8, 0x66, 0x2B, 0xFC, 0xB1, + 0xED, 0xA0, 0x77, 0x3A, 0x94, 0xD9, 0x0E, 0x43, + 0xB6, 0xFB, 0x2C, 0x61, 0xCF, 0x82, 0x55, 0x18, + 0x44, 0x09, 0xDE, 0x93, 0x3D, 0x70, 0xA7, 0xEA, + 0x3E, 0x73, 0xA4, 0xE9, 0x47, 0x0A, 0xDD, 0x90, + 0xCC, 0x81, 0x56, 0x1B, 0xB5, 0xF8, 0x2F, 0x62, + 0x97, 0xDA, 0x0D, 0x40, 0xEE, 0xA3, 0x74, 0x39, + 0x65, 0x28, 0xFF, 0xB2, 0x1C, 0x51, 0x86, 0xCB, + 0x21, 0x6C, 0xBB, 0xF6, 0x58, 0x15, 0xC2, 0x8F, + 0xD3, 0x9E, 0x49, 0x04, 0xAA, 0xE7, 0x30, 0x7D, + 0x88, 0xC5, 0x12, 0x5F, 0xF1, 0xBC, 0x6B, 0x26, + 0x7A, 0x37, 0xE0, 0xAD, 0x03, 0x4E, 0x99, 0xD4, + 0x7C, 0x31, 0xE6, 0xAB, 0x05, 0x48, 0x9F, 0xD2, + 0x8E, 0xC3, 0x14, 0x59, 0xF7, 0xBA, 0x6D, 0x20, + 0xD5, 0x98, 0x4F, 0x02, 0xAC, 0xE1, 0x36, 0x7B, + 0x27, 0x6A, 0xBD, 0xF0, 0x5E, 0x13, 0xC4, 0x89, + 0x63, 0x2E, 0xF9, 0xB4, 0x1A, 0x57, 0x80, 0xCD, + 0x91, 0xDC, 0x0B, 0x46, 0xE8, 0xA5, 0x72, 0x3F, + 0xCA, 0x87, 0x50, 0x1D, 0xB3, 0xFE, 0x29, 0x64, + 0x38, 0x75, 0xA2, 0xEF, 0x41, 0x0C, 0xDB, 0x96, + 0x42, 0x0F, 0xD8, 0x95, 0x3B, 0x76, 0xA1, 0xEC, + 0xB0, 0xFD, 0x2A, 0x67, 0xC9, 0x84, 0x53, 0x1E, + 0xEB, 0xA6, 0x71, 0x3C, 0x92, 0xDF, 0x08, 0x45, + 0x19, 0x54, 0x83, 0xCE, 0x60, 0x2D, 0xFA, 0xB7, + 0x5D, 0x10, 0xC7, 0x8A, 0x24, 0x69, 0xBE, 0xF3, + 0xAF, 0xE2, 0x35, 0x78, 0xD6, 0x9B, 0x4C, 0x01, + 0xF4, 0xB9, 0x6E, 0x23, 0x8D, 0xC0, 0x17, 0x5A, + 0x06, 0x4B, 0x9C, 0xD1, 0x7F, 0x32, 0xE5, 0xA8 +}; + +static int check_inpage_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + int ret = 0; + + if (book == TAS2781_YRAM_BOOK1) { + if (page == TAS2781_YRAM1_PAGE) { + if (reg >= TAS2781_YRAM1_START_REG) { + cd->offset = reg; + cd->len = len; + ret = 1; + } else if ((reg + len) > TAS2781_YRAM1_START_REG) { + cd->offset = TAS2781_YRAM1_START_REG; + cd->len = + len - (TAS2781_YRAM1_START_REG - reg); + ret = 1; + } else + ret = 0; + } else if (page == TAS2781_YRAM3_PAGE) { + if (reg > TAS2781_YRAM3_END_REG) { + ret = 0; + } else if (reg >= TAS2781_YRAM3_START_REG) { + if ((reg + len) > TAS2781_YRAM3_END_REG) { + cd->offset = reg; + cd->len = + TAS2781_YRAM3_END_REG - reg + 1; + ret = 1; + } else { + cd->offset = reg; + cd->len = len; + ret = 1; + } + } else { + if ((reg + (len-1)) < + TAS2781_YRAM3_START_REG) + ret = 0; + else { + cd->offset = + TAS2781_YRAM3_START_REG; + cd->len = + len - (TAS2781_YRAM3_START_REG - reg); + ret = 1; + } + } + } + } else if (book == + TAS2781_YRAM_BOOK2) { + if (page == TAS2781_YRAM5_PAGE) { + if (reg > TAS2781_YRAM5_END_REG) { + ret = 0; + } else if (reg >= TAS2781_YRAM5_START_REG) { + if ((reg + len) > TAS2781_YRAM5_END_REG) { + cd->offset = reg; + cd->len = + TAS2781_YRAM5_END_REG - reg + 1; + ret = 1; + } else { + cd->offset = reg; + cd->len = len; + ret = 1; + } + } else { + if ((reg + (len-1)) < + TAS2781_YRAM5_START_REG) + ret = 0; + else { + cd->offset = + TAS2781_YRAM5_START_REG; + cd->len = + len - (TAS2781_YRAM5_START_REG - reg); + ret = 1; + } + } + } + } else + ret = 0; + + return ret; +} + +static int check_inblock_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + int ret = 0; + + if (book == TAS2781_YRAM_BOOK1) { + if (page < TAS2781_YRAM2_START_PAGE) + ret = 0; + else if (page <= TAS2781_YRAM2_END_PAGE) { + if (reg > TAS2781_YRAM2_END_REG) + ret = 0; + else if (reg >= TAS2781_YRAM2_START_REG) { + cd->offset = reg; + cd->len = len; + ret = 1; + } else { + if ((reg + (len-1)) < + TAS2781_YRAM2_START_REG) + ret = 0; + else { + cd->offset = + TAS2781_YRAM2_START_REG; + cd->len = + reg + len - TAS2781_YRAM2_START_REG; + ret = 1; + } + } + } else + ret = 0; + } else if (book == + TAS2781_YRAM_BOOK2) { + if (page < TAS2781_YRAM4_START_PAGE) + ret = 0; + else if (page <= TAS2781_YRAM4_END_PAGE) { + if (reg > TAS2781_YRAM2_END_REG) + ret = 0; + else if (reg >= TAS2781_YRAM2_START_REG) { + cd->offset = reg; + cd->len = len; + ret = 1; + } else { + if ((reg + (len-1)) + < TAS2781_YRAM2_START_REG) + ret = 0; + else { + cd->offset = + TAS2781_YRAM2_START_REG; + cd->len = + reg + len - TAS2781_YRAM2_START_REG; + ret = 1; + } + } + } else + ret = 0; + } else + ret = 0; + + return ret; +} + +static int check_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + int ret = 0; + + ret = check_inpage_yram(cd, book, page, reg, len); + if (ret == 0) + ret = check_inblock_yram(cd, book, + page, reg, len); + + return ret; +} + +static int do_singlereg_checksum(struct tasdevice_priv *tasdevice, + enum channel chl, unsigned char book, unsigned char page, + unsigned char reg, unsigned char val) +{ + int ret = 0; + struct tas_crc crc_data; + unsigned int nData1 = 0; + + if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) + && (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) + && (reg >= TASDEVICE_PAGE_REG(TAS2781_SA_COEFF_SWAP_REG)) + && (reg <= (TASDEVICE_PAGE_REG( + TAS2781_SA_COEFF_SWAP_REG) + 4))) { + /*DSP swap command, pass */ + ret = 0; + goto end; + } + + ret = check_yram(&crc_data, book, page, reg, 1); + if (ret == 1) { + ret = tasdevice_dev_read(tasdevice, chl, + TASDEVICE_REG(book, page, reg), &nData1); + if (ret < 0) + goto end; + + if (nData1 != val) { + dev_err(tasdevice->dev, + "B[0x%x]P[0x%x]R[0x%x] W[0x%x], R[0x%x]\n", + book, page, reg, + val, nData1); + ret = -EAGAIN; + tasdevice->tasdevice[chl].err_code |= + ERROR_YRAM_CRCCHK; + ret = -EAGAIN; + goto end; + } + + ret = crc8(crc8_lookup_table, &val, 1, 0); + } + +end: + return ret; +} + +static int do_multireg_checksum(struct tasdevice_priv *tasdevice, + enum channel chn, unsigned char book, unsigned char page, + unsigned char reg, unsigned int len) +{ + int ret = 0, i = 0; + unsigned char crc_chksum = 0; + unsigned char nBuf1[128] = {0}; + struct tas_crc crc_data; + + if ((reg + len-1) > 127) { + ret = -EINVAL; + dev_err(tasdevice->dev, "firmware error\n"); + goto end; + } + + if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) + && (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) + && (reg == TASDEVICE_PAGE_REG(TAS2781_SA_COEFF_SWAP_REG)) + && (len == 4)) { + /*DSP swap command, pass */ + ret = 0; + goto end; + } + + ret = check_yram(&crc_data, book, page, reg, len); + dev_info(tasdevice->dev, + "isYRAM: Bk 0x%02x, Pg 0x%02x, Rg 0x%02x\n", + book, page, reg); + dev_info(tasdevice->dev, + "isYRAM: crc_data.mnLen 0x%x, len 0x%x, nResult %d\n", + crc_data.len, len, ret); + dev_info(tasdevice->dev, "crc_data.mnOffset %x\n", crc_data.offset); + if (ret == 1) { + if (len == 1) { + dev_err(tasdevice->dev, "firmware error\n"); + ret = -EINVAL; + goto end; + } else { + ret = tasdevice_dev_bulk_read(tasdevice, chn, + TASDEVICE_REG(book, page, crc_data.offset), + nBuf1, crc_data.len); + if (ret < 0) + goto end; + + for (i = 0; i < crc_data.len; i++) { + if ((book == TASDEVICE_BOOK_ID( + TAS2781_SA_COEFF_SWAP_REG)) + && (page == TASDEVICE_PAGE_ID( + TAS2781_SA_COEFF_SWAP_REG)) + && ((i + crc_data.offset) + >= TASDEVICE_PAGE_REG( + TAS2781_SA_COEFF_SWAP_REG)) + && ((i + crc_data.offset) + <= (TASDEVICE_PAGE_REG( + TAS2781_SA_COEFF_SWAP_REG) + + 4))) { + /*DSP swap command, bypass */ + continue; + } else + crc_chksum += + crc8(crc8_lookup_table, + &nBuf1[i], 1, 0); + } + + ret = crc_chksum; + } + } + +end: + return ret; +} + +static int tasdevice_load_block(struct tasdevice_priv *tas_dev, + struct tasdev_blk *block) +{ + int ret = 0; + unsigned int n_cmds = 0; + unsigned char book = 0; + unsigned char page = 0; + unsigned char offset = 0; + unsigned char val = 0; + unsigned int len = 0; + unsigned int sleep_dur = 0; + unsigned char crc_chksum = 0; + unsigned int n_value = 0; + int n_retry = 6; + unsigned char *data = block->data; + int chn = 0, chnend = 0; + + dev_info(tas_dev->dev, + "load block: Type = %d, commands = %d\n", + block->type, block->n_cmds); + switch (block->type) { + case MAIN_ALL_DEVICES: + chn = 0; + chnend = tas_dev->ndev; + break; + case MAIN_DEVICE_A: + case COEFF_DEVICE_A: + case PRE_DEVICE_A: + chn = 0; + chnend = 1; + break; + case MAIN_DEVICE_B: + case COEFF_DEVICE_B: + case PRE_DEVICE_B: + chn = 1; + chnend = 2; + break; + case MAIN_DEVICE_C: + case COEFF_DEVICE_C: + case PRE_DEVICE_C: + chn = 2; + chnend = 3; + break; + case MAIN_DEVICE_D: + case COEFF_DEVICE_D: + case PRE_DEVICE_D: + chn = 3; + chnend = 4; + break; + default: + dev_info(tas_dev->dev, + "load block: Other Type = 0x%02x\n", + block->type); + break; + } + + for (; chn < chnend; chn++) { + if (tas_dev->tasdevice[chn].is_loading == false) + continue; +start: + if (block->is_pchksum_present) { + ret = tasdevice_dev_write(tas_dev, chn, + TASDEVICE_I2CChecksum, 0); + if (ret < 0) + goto end; + } + + if (block->is_ychksum_present) + crc_chksum = 0; + + n_cmds = 0; + + while (n_cmds < block->n_cmds) { + data = block->data + n_cmds * 4; + + book = data[0]; + page = data[1]; + offset = data[2]; + val = data[3]; + + n_cmds++; + + if (offset <= 0x7F) { + ret = tasdevice_dev_write(tas_dev, chn, + TASDEVICE_REG(book, page, offset), + val); + if (ret < 0) + goto end; + if (block->is_ychksum_present) { + ret = do_singlereg_checksum(tas_dev, + chn, book, page, offset, + val); + if (ret < 0) + goto check; + crc_chksum += (unsigned char)ret; + } + continue; + } + if (offset == 0x81) { + sleep_dur = (book << 8) + page; + msleep(sleep_dur); + continue; + } + if (offset == 0x85) { + data += 4; + len = (book << 8) + page; + book = data[0]; + page = data[1]; + offset = data[2]; + if (len > 1) { + ret = tasdevice_dev_bulk_write( + tas_dev, chn, + TASDEVICE_REG(book, page, + offset), data + 3, len); + if (ret < 0) + goto end; + if (!block->is_ychksum_present) + goto next; + ret = do_multireg_checksum( + tas_dev, chn, book, + page, offset, + len); + if (ret < 0) + goto check; + crc_chksum += + (unsigned char)ret; + } else { + ret = tasdevice_dev_write(tas_dev, + chn, + TASDEVICE_REG(book, page, + offset), + data[3]); + if (ret < 0) + goto end; + if (!block->is_ychksum_present) + goto next; + ret = do_singlereg_checksum( + tas_dev, chn, book, + page, offset, + data[3]); + if (ret < 0) + goto check; + crc_chksum += + (unsigned char)ret; + } +next: + n_cmds++; + if (len >= 2) + n_cmds += ((len - 2) / 4) + 1; + } + } + if (block->is_pchksum_present) { + ret = tasdevice_dev_read(tas_dev, chn, + TASDEVICE_I2CChecksum, &n_value); + if (ret < 0) { + dev_err(tas_dev->dev, "%s: Channel %d\n", + __func__, chn); + goto check; + } + if ((n_value & 0xff) != block->pchksum) { + dev_err(tas_dev->dev, + "Block PChkSum Channel %d ", chn); + dev_err(tas_dev->dev, + "FW = 0x%x, Reg = 0x%x\n", + block->pchksum, + (n_value & 0xff)); + ret = -EAGAIN; + tas_dev->tasdevice[chn].err_code |= + ERROR_PRAM_CRCCHK; + goto check; + } + ret = 0; + tas_dev->tasdevice[chn].err_code &= + ~ERROR_PRAM_CRCCHK; + dev_info(tas_dev->dev, "Block[0x%02x] PChkSum match\n", + block->type); + } + + if (block->is_ychksum_present) { + //TBD, open it when FW ready + dev_err(tas_dev->dev, + "Block YChkSum: FW = 0x%x, YCRC = 0x%x\n", + block->ychksum, crc_chksum); + + tas_dev->tasdevice[chn].err_code &= + ~ERROR_YRAM_CRCCHK; + ret = 0; + dev_info(tas_dev->dev, + "Block[0x%x] YChkSum match\n", block->type); + } +check: + if (ret == -EAGAIN) { + n_retry--; + if (n_retry > 0) + goto start; + else { + if ((block->type == MAIN_ALL_DEVICES) + || (block->type == MAIN_DEVICE_A) + || (block->type == MAIN_DEVICE_B) + || (block->type == MAIN_DEVICE_C) + || (block->type == MAIN_DEVICE_D)) + tas_dev->tasdevice[chn].cur_prog = -1; + else + tas_dev->tasdevice[chn].cur_conf = -1; + n_retry = 6; + } + } + } +end: + if (ret < 0) + dev_err(tas_dev->dev, "Block (%d) load error\n", block->type); + + return ret; +} + + +static int tasdevice_load_data(struct tasdevice_priv *tas_dev, + struct tasdevice_data *dev_data) +{ + int ret = 0; + unsigned int n_block = 0; + struct tasdev_blk *block = NULL; + + dev_info(tas_dev->dev, "%s: TAS2781 load data: %s, Blocks = %d\n", + __func__, dev_data->name, dev_data->n_blk); + + for (n_block = 0; n_block < dev_data->n_blk; n_block++) { + block = &(dev_data->dev_blks[n_block]); + ret = tas_dev->tasdevice_load_block(tas_dev, block); + if (ret < 0) + break; + } + + return ret; +} + +static int tasdevice_load_calibrated_data( + struct tasdevice_priv *tas_dev, struct tasdevice_data *dev_data) +{ + int ret = 0; + unsigned int n_block = 0; + struct tasdev_blk *block = NULL; + + dev_info(tas_dev->dev, "%s: TAS2781 load data: %s, Blocks = %d\n", + __func__, dev_data->name, dev_data->n_blk); + + for (n_block = 0; n_block < dev_data->n_blk; n_block++) { + block = &(dev_data->dev_blks[n_block]); + ret = tasdevice_load_block(tas_dev, block); + if (ret < 0) + break; + } + + return ret; +} + +int tas2781_load_calibration(void *context, + char *file_name, enum channel i) +{ + int ret = 0, offset = 0; + struct firmware fmw; + const struct firmware *fw_entry = NULL; + struct tasdevice_priv *tas_dev = (struct tasdevice_priv *)context; + struct tasdevice *tasdev = &(tas_dev->tasdevice[i]); + struct tasdevice_fw *tas_fmw = NULL; + + ret = request_firmware(&fw_entry, file_name, tas_dev->dev); + if (!ret) { + if (!fw_entry->size) { + dev_err(tas_dev->dev, + "%s: file read error: size = %lu\n", + __func__, (unsigned long)fw_entry->size); + goto out; + } + fmw.size = fw_entry->size; + fmw.data = fw_entry->data; + dev_info(tas_dev->dev, + "%s: file = %s, file size %zd\n", + __func__, file_name, fw_entry->size); + } else { + dev_info(tas_dev->dev, + "%s: Request firmware %s failed\n", + __func__, file_name); + goto out; + } + + tas_fmw = tasdev->cali_data_fmw = kcalloc(1, + sizeof(struct tasdevice_fw), GFP_KERNEL); + if (tasdev->cali_data_fmw == NULL) { + dev_err(tas_dev->dev, "%s: FW memory failed!\n", __func__); + ret = -1; + goto out; + } + tas_fmw->dev = tas_dev->dev; + offset = fw_parse_header(tas_dev, tas_fmw, &fmw, offset); + if (offset == -1) { + dev_err(tas_dev->dev, "%s: fw_parse_header EXIT!\n", __func__); + goto out; + } + offset = fw_parse_variable_hdr_cal(tas_dev, tas_fmw, &fmw, offset); + if (offset == -1) { + dev_err(tas_dev->dev, + "%s: fw_parse_variable_header_cal EXIT!\n", __func__); + goto out; + } + offset = fw_parse_program_data(tas_dev, tas_fmw, &fmw, offset); + if (offset == -1) { + dev_err(tas_dev->dev, "%s: fw_parse_program_data EXIT!\n", + __func__); + goto out; + } + offset = fw_parse_configuration_data(tas_dev, tas_fmw, &fmw, + offset); + if (offset == -1) { + dev_err(tas_dev->dev, + "%s: fw_parse_configuration_data EXIT!\n", __func__); + goto out; + } + offset = fw_parse_calibration_data(tas_dev, + tas_fmw, &fmw, offset); + if (offset == -1) { + dev_err(tas_dev->dev, "%s: fw_parse_calibration_data EXIT!\n", + __func__); + goto out; + } + tasdev->is_calibrated_data_loaded = true; +out: + if (fw_entry) { + release_firmware(fw_entry); + fw_entry = NULL; + } + return ret; +} + +int tasdevice_dspfw_ready(const struct firmware *fmw, void *context) +{ + struct tasdevice_priv *tas_dev = (struct tasdevice_priv *) context; + struct tasdevice_fw *tas_fmw = NULL; + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr = NULL; + int offset = 0, ret = 0; + + if (!fmw || !fmw->data) { + dev_err(tas_dev->dev, "%s: Failed to read firmware %s\n", + __func__, tas_dev->coef_binaryname); + ret = -1; + goto out; + } + + tas_dev->fmw = kcalloc(1, + sizeof(struct tasdevice_fw), GFP_KERNEL); + if (tas_dev->fmw == NULL) { + ret = -1; + goto out; + } + tas_fmw = tas_dev->fmw; + tas_fmw->dev = tas_dev->dev; + offset = fw_parse_header(tas_dev, tas_fmw, fmw, offset); + + if (offset == -1) + goto out; + fw_fixed_hdr = &(tas_fmw->fw_hdr.fixed_hdr); + switch (fw_fixed_hdr->drv_ver) { + case 0x301: + case 0x302: + case 0x502: + tas_dev->fw_parse_variable_header = + fw_parse_variable_header_kernel; + tas_dev->fw_parse_program_data = + fw_parse_program_data_kernel; + tas_dev->fw_parse_configuration_data = + fw_parse_configuration_data_kernel; + tas_dev->tasdevice_load_block = + tasdevice_load_block_kernel; + break; + case 0x202: + case 0x400: + tas_dev->fw_parse_variable_header = + fw_parse_variable_header_git; + tas_dev->fw_parse_program_data = + fw_parse_program_data; + tas_dev->fw_parse_configuration_data = + fw_parse_configuration_data; + tas_dev->tasdevice_load_block = + tasdevice_load_block; + break; + default: + if (fw_fixed_hdr->drv_ver == 0x100) { + if (fw_fixed_hdr->ppcver >= PPC3_VERSION) { + tas_dev->fw_parse_variable_header = + fw_parse_variable_header_kernel; + tas_dev->fw_parse_program_data = + fw_parse_program_data_kernel; + tas_dev->fw_parse_configuration_data = + fw_parse_configuration_data_kernel; + tas_dev->tasdevice_load_block = + tasdevice_load_block_kernel; + tas_dev->fw_parse_calibration_data = NULL; + } else { + switch (fw_fixed_hdr->ppcver) { + case 0x00: + tas_dev->fw_parse_variable_header = + fw_parse_variable_header_git; + tas_dev->fw_parse_program_data = + fw_parse_program_data; + tas_dev->fw_parse_configuration_data = + fw_parse_configuration_data; + tas_dev->fw_parse_calibration_data = + fw_parse_calibration_data; + tas_dev->tasdevice_load_block = + tasdevice_load_block; + break; + default: + dev_err(tas_dev->dev, + "%s: PPCVersion must be 0x0 or 0x%02x", + __func__, PPC3_VERSION); + dev_err(tas_dev->dev, " Current:0x%02x\n", + fw_fixed_hdr->ppcver); + offset = -1; + break; + } + } + } else { + dev_err(tas_dev->dev, + "%s: DriverVersion must be 0x0, 0x230 or above 0x230 ", + __func__); + dev_err(tas_dev->dev, "current is 0x%02x\n", + fw_fixed_hdr->drv_ver); + offset = -1; + } + break; + } + + offset = tas_dev->fw_parse_variable_header(tas_dev, fmw, offset); + if (offset == -1) + goto out; + + offset = tas_dev->fw_parse_program_data(tas_dev, tas_fmw, fmw, + offset); + if (offset < 0) { + ret = -1; + goto out; + } + offset = tas_dev->fw_parse_configuration_data(tas_dev, + tas_fmw, fmw, offset); + if (offset < 0) + ret = -1; + +out: + return ret; +} + +void tasdevice_calbin_remove(void *context) +{ + struct tasdevice_priv *tas_dev = (struct tasdevice_priv *) context; + struct tasdevice *tasdev = NULL; + int i = 0; + + if (tas_dev) { + for (i = 0; i < tas_dev->ndev; i++) { + tasdev = &(tas_dev->tasdevice[i]); + if (tasdev->cali_data_fmw) { + tas2781_clear_calfirmware( + tasdev->cali_data_fmw); + tasdev->cali_data_fmw = NULL; + } + } + } +} + +void tasdevice_dsp_remove(void *context) +{ + struct tasdevice_priv *tas_dev = (struct tasdevice_priv *) context; + int i; + struct tasdev_blk *blk; + unsigned int n_blk; + + if (tas_dev->fmw) { + struct tasdevice_fw *tas_fmw = tas_dev->fmw; + + if (tas_fmw->programs) { + struct tasdevice_prog *program; + + for (i = 0; i < tas_fmw->n_programs; i++) { + program = &(tas_fmw->programs[i]); + if (program) { + struct tasdevice_data + *im = &(program->dev_data); + + if (!im->dev_blks) + continue; + + for (n_blk = 0; n_blk < im->n_blk; + n_blk++) { + blk = + &(im->dev_blks[n_blk]); + kfree(blk->data); + } + kfree(im->dev_blks); + } + } + kfree(tas_fmw->programs); + } + + if (tas_fmw->configs) { + struct tasdevice_config *config; + + for (i = 0; i < tas_fmw->n_configurations; i++) { + config = &(tas_fmw->configs[i]); + if (config) { + struct tasdevice_data + *im = &(config->dev_data); + + if (!im->dev_blks) + continue; + for (n_blk = 0; n_blk < im->n_blk; + n_blk++) { + blk = + &(im->dev_blks[n_blk]); + kfree(blk->data); + } + kfree(im->dev_blks); + } + } + kfree(tas_fmw->configs); + } + kfree(tas_fmw); + tas_dev->fmw = NULL; + } +} + +static int tas2781_set_calibration(void *context, enum channel i, + int n_calibration) +{ + struct tasdevice_priv *tas_dev = (struct tasdevice_priv *) context; + int ret = 0; + struct tasdevice *tasdevice = &(tas_dev->tasdevice[i]); + struct tasdevice_fw *cal_fmw = tasdevice->cali_data_fmw; + + if ((!tas_dev->fmw->programs) + || (!tas_dev->fmw->configs)) { + dev_err(tas_dev->dev, "%s, Firmware not loaded\n\r", __func__); + ret = 0; + goto out; + } + + if (n_calibration == 0xFF || (n_calibration == 0x100 + && tasdevice->is_calibrated_data_loaded == false)) { + if (cal_fmw) { + tasdevice->is_calibrated_data_loaded = false; + tas2781_clear_calfirmware(cal_fmw); + cal_fmw = NULL; + } + + scnprintf(tas_dev->cal_binaryname[i], 64, "%s_cal_0x%02x.bin", + tas_dev->dev_name, tas_dev->tasdevice[i].dev_addr); + ret = tas2781_load_calibration(tas_dev, + tas_dev->cal_binaryname[i], i); + if (ret != 0) { + dev_err(tas_dev->dev, + "%s: load %s error, no-side effect\n", + __func__, tas_dev->cal_binaryname[i]); + ret = 0; + } + } + tasdevice->is_loading = true; + tasdevice->is_loaderr = false; + + if (cal_fmw) { + struct tasdevice_calibration *calibration = + cal_fmw->calibrations; + + if (calibration) + tasdevice_load_calibrated_data(tas_dev, + &(calibration->dev_data)); + } else + dev_err(tas_dev->dev, + "%s: No calibrated data for device %d\n", __func__, i); + +out: + return ret; +} + +int tasdevice_select_tuningprm_cfg(void *context, int prm_no, + int cfg_no, int rca_conf_no) +{ + struct tasdevice_priv *tas_dev = (struct tasdevice_priv *) context; + struct tasdevice_rca *rca = &(tas_dev->rcabin); + struct tasdevice_config_info **cfg_info = rca->cfg_info; + struct tasdevice_fw *tas_fmw = tas_dev->fmw; + struct tasdevice_config *conf = NULL; + struct tasdevice_prog *program = NULL; + int i = 0; + int status = 0; + int prog_status = 0; + + if (tas_fmw == NULL) { + dev_err(tas_dev->dev, "%s: Firmware is NULL\n", __func__); + goto out; + } + + if (cfg_no >= tas_fmw->n_configurations) { + dev_err(tas_dev->dev, + "%s: cfg(%d) is not in range of conf %u\n", + __func__, cfg_no, tas_fmw->n_configurations); + goto out; + } + + if (prm_no >= tas_fmw->n_programs) { + dev_err(tas_dev->dev, + "%s: prm(%d) is not in range of Programs %u\n", + __func__, prm_no, tas_fmw->n_programs); + goto out; + } + + if (rca_conf_no > rca->ncfgs || rca_conf_no <= 0 || + cfg_info == NULL) { + dev_err(tas_dev->dev, + "conf_no:%d should be in range from 0 to %u\n", + rca_conf_no, rca->ncfgs-1); + goto out; + } else { + dev_info(tas_dev->dev, "%s: rca_profile_conf_id = %d\n", + __func__, rca_conf_no); + } + + conf = &(tas_fmw->configs[cfg_no]); + for (i = 0; i < tas_dev->ndev; i++) { + if (cfg_info[rca_conf_no]->active_dev & (1 << i)) { + if (tas_dev->tasdevice[i].cur_prog != prm_no) { + tas_dev->tasdevice[i].cur_conf = -1; + tas_dev->tasdevice[i].is_loading = true; + prog_status++; + } + } else + tas_dev->tasdevice[i].is_loading = false; + tas_dev->tasdevice[i].is_loaderr = false; + } + + if (prog_status) { + program = &(tas_fmw->programs[prm_no]); + tasdevice_load_data(tas_dev, &(program->dev_data)); + for (i = 0; i < tas_dev->ndev; i++) { + if (tas_dev->tasdevice[i].is_loaderr == true) + continue; + else if (tas_dev->tasdevice[i].is_loaderr == false + && tas_dev->tasdevice[i].is_loading == true) { + struct tasdevice_fw *cal_fmw = + tas_dev->tasdevice[i].cali_data_fmw; + + if (cal_fmw) { + struct tasdevice_calibration + *calibration = + cal_fmw->calibrations; + + if (calibration) + tasdevice_load_calibrated_data( + tas_dev, + &(calibration->dev_data)); + } + tas_dev->tasdevice[i].cur_prog = prm_no; + } + } + } + + if (tas_dev->is_glb_calibrated_data_loaded == false) { + for (i = 0; i < tas_dev->ndev; i++) + tas2781_set_calibration(tas_dev, i, 0x100); + tas_dev->is_glb_calibrated_data_loaded = true; + /* No wise to reload calibrationdata everytime, + * this code will work once even if calibrated + * data still failed to be got + */ + } + + for (i = 0; i < tas_dev->ndev; i++) { + dev_info(tas_dev->dev, "%s,fun %d,%d,%d\n", __func__, + tas_dev->tasdevice[i].cur_conf, + cfg_info[rca_conf_no]->active_dev, + tas_dev->tasdevice[i].is_loaderr); + if (tas_dev->tasdevice[i].cur_conf != cfg_no + && (cfg_info[rca_conf_no]->active_dev & (1 << i)) + && (tas_dev->tasdevice[i].is_loaderr == false)) { + status++; + tas_dev->tasdevice[i].is_loading = true; + } else + tas_dev->tasdevice[i].is_loading = false; + } + + if (status) { + status = 0; + tasdevice_load_data(tas_dev, &(conf->dev_data)); + for (i = 0; i < tas_dev->ndev; i++) { + if (tas_dev->tasdevice[i].is_loaderr == true) { + status |= 1 << (i + 4); + continue; + } else if (tas_dev->tasdevice[i].is_loaderr == false + && tas_dev->tasdevice[i].is_loading == true) + tas_dev->tasdevice[i].cur_conf = cfg_no; + } + } else { + dev_err(tas_dev->dev, + "%s: No device is in active in conf %d\n", + __func__, rca_conf_no); + } + + status |= cfg_info[rca_conf_no]->active_dev; + dev_info(tas_dev->dev, "%s: DSP mode: load status is %08x\n", + __func__, status); +out: + return prog_status; +} diff --git a/sound/soc/codecs/tas2781-dsp.h b/sound/soc/codecs/tas2781-dsp.h new file mode 100644 index 0000000..639e84e --- /dev/null +++ b/sound/soc/codecs/tas2781-dsp.h @@ -0,0 +1,178 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier + * + * Copyright (C) 2022 - 2023 Texas Instruments Incorporated + * https://www.ti.com + * + * The TAS2781 driver implements a flexible and configurable algo coff setting + * for one, two, even multiple TAS2781 chips. + * + * Author: Shenghao Ding <shenghao-ding(a)ti.com> + * Kevin Lu <kevin-lu(a)ti.com> + */ + +#ifndef __TASDEVICE_DSP_H__ +#define __TASDEVICE_DSP_H__ + +#define MAIN_ALL_DEVICES 0x0d +#define MAIN_DEVICE_A 0x01 +#define MAIN_DEVICE_B 0x08 +#define MAIN_DEVICE_C 0x10 +#define MAIN_DEVICE_D 0x14 +#define COEFF_DEVICE_A 0x03 +#define COEFF_DEVICE_B 0x0a +#define COEFF_DEVICE_C 0x11 +#define COEFF_DEVICE_D 0x15 +#define PRE_DEVICE_A 0x04 +#define PRE_DEVICE_B 0x0b +#define PRE_DEVICE_C 0x12 +#define PRE_DEVICE_D 0x16 + +#define PPC3_VERSION 0x4100 +#define RCA_CONFIGID_BYPASS_ALL 0 +#define TASDEVICE_DEVICE_SUM 8 +#define TASDEVICE_CONFIG_SUM 64 + +enum channel { + top_left_Chn, + top_right_chn, + bottom_left_Chn, + bottom_right_chn, + max_chn, +}; + +enum tasdevice_dsp_dev_idx { + TASDEVICE_DSP_TAS_2555 = 0, + TASDEVICE_DSP_TAS_2555_STEREO, + TASDEVICE_DSP_TAS_2557_MONO, + TASDEVICE_DSP_TAS_2557_DUAL_MONO, + TASDEVICE_DSP_TAS_2559, + TASDEVICE_DSP_TAS_2563, + TASDEVICE_DSP_TAS_2563_DUAL_MONO = 7, + TASDEVICE_DSP_TAS_2563_QUAD, + TASDEVICE_DSP_TAS_2563_21, + TASDEVICE_DSP_TAS_2781, + TASDEVICE_DSP_TAS_2781_DUAL_MONO, + TASDEVICE_DSP_TAS_2781_21, + TASDEVICE_DSP_TAS_2781_QUAD, + TASDEVICE_DSP_TAS_MAX_DEVICE +}; + +struct tasdevice_fw_fixed_hdr { + unsigned int fwsize; + unsigned int ppcver; + unsigned int drv_ver; +}; + +struct tasdevice_dspfw_hdr { + struct tasdevice_fw_fixed_hdr fixed_hdr; + unsigned short device_family; + unsigned short device; + unsigned char ndev; +}; + +struct tasdev_blk { + unsigned int type; + unsigned char is_pchksum_present; + unsigned char pchksum; + unsigned char is_ychksum_present; + unsigned char ychksum; + unsigned int n_cmds; + unsigned int blk_size; + unsigned int n_subblocks; + unsigned char *data; +}; + +struct tasdevice_data { + char name[64]; + unsigned int n_blk; + struct tasdev_blk *dev_blks; +}; +struct tasdevice_prog { + unsigned int prog_size; + struct tasdevice_data dev_data; +}; + +struct tasdevice_config { + unsigned int cfg_size; + char name[64]; + struct tasdevice_data dev_data; +}; + +struct tasdevice_calibration { + struct tasdevice_data dev_data; +}; + +struct tasdevice_fw { + struct tasdevice_dspfw_hdr fw_hdr; + unsigned short n_programs; + struct tasdevice_prog *programs; + unsigned short n_configurations; + struct tasdevice_config *configs; + unsigned short n_calibrations; + struct tasdevice_calibration *calibrations; + struct device *dev; +}; + +enum tasdevice_dsp_fw_state { + TASDEVICE_DSP_FW_NONE = 0, + TASDEVICE_DSP_FW_PENDING, + TASDEVICE_DSP_FW_FAIL, + TASDEVICE_DSP_FW_ALL_OK, +}; + +enum tasdevice_bin_blk_type { + TASDEVICE_BIN_BLK_COEFF = 1, + TASDEVICE_BIN_BLK_POST_POWER_UP, + TASDEVICE_BIN_BLK_PRE_SHUTDOWN, + TASDEVICE_BIN_BLK_PRE_POWER_UP, + TASDEVICE_BIN_BLK_POST_SHUTDOWN +}; + +struct tasdevice_rca_hdr { + unsigned int img_sz; + unsigned int checksum; + unsigned int binary_version_num; + unsigned int drv_fw_version; + unsigned char plat_type; + unsigned char dev_family; + unsigned char reserve; + unsigned char ndev; + unsigned char devs[TASDEVICE_DEVICE_SUM]; + unsigned int nconfig; + unsigned int config_size[TASDEVICE_CONFIG_SUM]; +}; + +struct tasdev_blk_data { + unsigned char dev_idx; + unsigned char block_type; + unsigned short yram_checksum; + unsigned int block_size; + unsigned int n_subblks; + unsigned char *regdata; +}; + +struct tasdevice_config_info { + unsigned int nblocks; + unsigned int real_nblocks; + unsigned char active_dev; + struct tasdev_blk_data **blk_data; +}; + +struct tasdevice_rca { + struct tasdevice_rca_hdr fw_hdr; + int ncfgs; + struct tasdevice_config_info **cfg_info; + int profile_cfg_id; +}; + +int tasdevice_dspfw_ready(const struct firmware *fmw, void *context); +void tasdevice_dsp_remove(void *context); +void tasdevice_calbin_remove(void *context); +int tas2781_load_calibration(void *tas_dev, char *filename, + enum channel i); +int tasdevice_select_tuningprm_cfg(void *context, int prm, + int cfg_no, int rca_conf_no); + +#endif diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c new file mode 100644 index 0000000..2b2b836 --- /dev/null +++ b/sound/soc/codecs/tas2781-i2c.c @@ -0,0 +1,1913 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier + * + * Copyright (C) 2022 - 2023 Texas Instruments Incorporated + * https://www.ti.com + * + * The TAS2781 driver implements a flexible and configurable algo coff setting + * for one, two, even multiple TAS2781 chips. + * + * Author: Shenghao Ding <shenghao-ding(a)ti.com> + * Kevin Lu <kevin-lu(a)ti.com> + */ + +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/init.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/of.h> +#include <linux/slab.h> +#include <linux/firmware.h> +#include <linux/of_gpio.h> +#include <linux/interrupt.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> + +#include "tas2781.h" + +/* max. length of the alsa mixer control name */ +#define MAX_CONTROL_NAME (48) +#define TASDEVICE_CLK_DIR_IN (0) +#define TASDEVICE_CLK_DIR_OUT (1) + +#define TASDEVICE_IRQ_DET_TIMEOUT (30000) +#define TASDEVICE_IRQ_DET_CNT_LIMIT (500) +#define TAS2781_REG_INT_LTCH0 TASDEVICE_REG(0X0, 0x0, 0x49) +#define TAS2781_REG_INT_LTCH1 TASDEVICE_REG(0X0, 0x0, 0x4A) +#define TAS2781_REG_INT_LTCH1_0 TASDEVICE_REG(0X0, 0x0, 0x4B) +#define TAS2781_REG_INT_LTCH2 TASDEVICE_REG(0X0, 0x0, 0x4F) +#define TAS2781_REG_INT_LTCH3 TASDEVICE_REG(0X0, 0x0, 0x50) +#define TAS2781_REG_INT_LTCH4 TASDEVICE_REG(0X0, 0x0, 0x51) + +const char *blocktype[5] = { + "COEFF", + "POST_POWER_UP", + "PRE_SHUTDOWN", + "PRE_POWER_UP", + "POST_SHUTDOWN" +}; + +static const char * const dts_dev_addr_tag[] = { + "topleft-channel", + "topright-channel", + "bottomleft-channel", + "bottomright-channel" +}; + +static const struct i2c_device_id tasdevice_id[] = { + { "tas2781", TAS2781 }, + {} +}; + +MODULE_DEVICE_TABLE(i2c, tasdevice_id); + +static const struct regmap_range_cfg tasdevice_ranges[] = { + { + .range_min = 0, + .range_max = 256 * 128, + .selector_reg = TASDEVICE_PAGE_SELECT, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = 128, + }, +}; + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 8, + .val_bits = 8, + .cache_type = REGCACHE_RBTREE, + .ranges = tasdevice_ranges, + .num_ranges = ARRAY_SIZE(tasdevice_ranges), + .max_register = 256 * 128, +}; + +static const struct of_device_id tasdevice_of_match[] = { + { .compatible = "ti,tas2781" }, + {}, +}; + +MODULE_DEVICE_TABLE(of, tasdevice_of_match); + +int tasdevice_process_block(void *context, + unsigned char *data, unsigned char dev_idx, int sublocksize) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *)context; + unsigned char subblk_typ = data[1]; + int subblk_offset = 2; + int chn = 0, chnend = 0; + int rc = 0; + int blktyp = dev_idx & 0xC0, idx = dev_idx & 0x3F; + bool is_err = false; + + if (idx) { + chn = idx-1; + chnend = idx; + } else { + if (tas_dev->set_global_mode) { + chn = tas_dev->ndev; + chnend = tas_dev->ndev + 1; + } else { + chn = 0; + chnend = tas_dev->ndev; + } + } + + for (; chn < chnend; chn++) { + if (tas_dev->set_global_mode == NULL && + tas_dev->tasdevice[chn].is_loading == false) + continue; + + is_err = false; + subblk_offset = 2; + switch (subblk_typ) { + case TASDEVICE_CMD_SING_W: { + int i = 0; + unsigned short len = SMS_HTONS(data[2], data[3]); + + subblk_offset += 2; + if (subblk_offset + 4 * len > sublocksize) { + dev_err(tas_dev->dev, + "process_block: Out of memory\n"); + is_err = true; + break; + } + + for (i = 0; i < len; i++) { + rc = tasdevice_dev_write(tas_dev, chn, + TASDEVICE_REG(data[subblk_offset], + data[subblk_offset + 1], + data[subblk_offset + 2]), + data[subblk_offset + 3]); + if (rc < 0) { + is_err = true; + dev_err(tas_dev->dev, + "process_block: single write error\n"); + } + subblk_offset += 4; + } + } + break; + case TASDEVICE_CMD_BURST: { + unsigned short len = + SMS_HTONS(data[2], data[3]); + + subblk_offset += 2; + if (subblk_offset + 4 + len > sublocksize) { + dev_err(tas_dev->dev, + "%s: BURST Out of memory\n", __func__); + is_err = true; + break; + } + if (len % 4) { + dev_err(tas_dev->dev, + "%s:Bst-len(%u)not div by 4\n", + __func__, len); + break; + } + + rc = tasdevice_dev_bulk_write(tas_dev, chn, + TASDEVICE_REG(data[subblk_offset], + data[subblk_offset + 1], + data[subblk_offset + 2]), + &(data[subblk_offset + 4]), + len); + if (rc < 0) { + is_err = true; + dev_err(tas_dev->dev, + "%s: bulk_write error = %d\n", + __func__, rc); + } + subblk_offset += (len + 4); + } + break; + case TASDEVICE_CMD_DELAY: { + unsigned short delay_time = 0; + + if (subblk_offset + 2 > sublocksize) { + dev_err(tas_dev->dev, + "%s: deley Out of memory\n", + __func__); + is_err = true; + break; + } + delay_time = SMS_HTONS(data[2], data[3]); + usleep_range(delay_time*1000, delay_time*1000); + subblk_offset += 2; + } + break; + case TASDEVICE_CMD_FIELD_W: + if (subblk_offset + 6 > sublocksize) { + dev_err(tas_dev->dev, + "%s: bit write Out of memory\n", + __func__); + is_err = true; + break; + } + rc = tasdevice_dev_update_bits(tas_dev, chn, + TASDEVICE_REG(data[subblk_offset + 2], + data[subblk_offset + 3], + data[subblk_offset + 4]), + data[subblk_offset + 1], + data[subblk_offset + 5]); + if (rc < 0) { + is_err = true; + dev_err(tas_dev->dev, + "%s: update_bits error = %d\n", + __func__, rc); + } + subblk_offset += 6; + break; + default: + break; + }; + if (is_err == true && blktyp != 0) { + if (blktyp == 0x80) { + tas_dev->tasdevice[chn].cur_prog = -1; + tas_dev->tasdevice[chn].cur_conf = -1; + } else + tas_dev->tasdevice[chn].cur_conf = -1; + } + } + return subblk_offset; +} + +static void tasdevice_select_cfg_blk(void *pContext, int conf_no, + unsigned char block_type) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) pContext; + struct tasdevice_rca *rca = &(tas_dev->rcabin); + struct tasdevice_config_info **cfg_info = rca->cfg_info; + int j = 0, k = 0, chn = 0, chnend = 0; + + if (conf_no >= rca->ncfgs || conf_no < 0 || NULL == cfg_info) { + dev_err(tas_dev->dev, + "conf_no should be not more than %u\n", + rca->ncfgs); + goto out; + } else + dev_info(tas_dev->dev, + "select_cfg_blk: profile_conf_id = %d\n", + conf_no); + + for (j = 0; j < (int)cfg_info[conf_no]->real_nblocks; j++) { + unsigned int length = 0, rc = 0; + + if (block_type > 5 || block_type < 2) { + dev_err(tas_dev->dev, + "ERROR!!!block_type should be in range from 2 to 5\n"); + goto out; + } + if (block_type != cfg_info[conf_no]->blk_data[j]->block_type) + continue; + dev_info(tas_dev->dev, + "%s: conf %d, block type:%s\t device idx = 0x%02x\n", + __func__, conf_no, + blocktype[cfg_info[conf_no]->blk_data[j] + ->block_type-1], cfg_info[conf_no]->blk_data[j] + ->dev_idx); + + for (k = 0; k < (int)cfg_info[conf_no]->blk_data[j] + ->n_subblks; k++) { + if (cfg_info[conf_no]->blk_data[j]->dev_idx) { + chn = + cfg_info[conf_no]->blk_data[j]->dev_idx + - 1; + chnend = + cfg_info[conf_no]->blk_data[j]->dev_idx; + } else { + chn = 0; + chnend = tas_dev->ndev; + } + for (; chn < chnend; chn++) + tas_dev->tasdevice[chn].is_loading = true; + + rc = tasdevice_process_block(tas_dev, + cfg_info[conf_no]->blk_data[j]->regdata + + length, + cfg_info[conf_no]->blk_data[j]->dev_idx, + cfg_info[conf_no]->blk_data[j]->block_size - + length); + length += rc; + if (cfg_info[conf_no]->blk_data[j]->block_size < + length) { + dev_err(tas_dev->dev, + "%s: ERROR:%u %u out of memory\n", + __func__, length, + cfg_info[conf_no]->blk_data[j] + ->block_size); + break; + } + } + if (length != cfg_info[conf_no]->blk_data[j]->block_size) + dev_err(tas_dev->dev, + "%s: %u %u size is not same\n", + __func__, length, + cfg_info[conf_no]->blk_data[j]->block_size); + + } + +out: + return; +} + +static struct tasdevice_config_info *tasdevice_add_config( + void *pContext, unsigned char *config_data, + unsigned int config_size) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *)pContext; + struct tasdevice_config_info *cfg_info = NULL; + int config_offset = 0, i = 0; + + cfg_info = kzalloc( + sizeof(struct tasdevice_config_info), GFP_KERNEL); + if (!cfg_info) + goto out; + + if (tas_dev->rcabin.fw_hdr.binary_version_num >= 0x105) { + if (config_offset + 64 > (int)config_size) { + dev_err(tas_dev->dev, + "add config: Out of memory\n"); + goto out; + } + config_offset += 64; + } + + if (config_offset + 4 > (int)config_size) { + dev_err(tas_dev->dev, + "add config: Out of memory\n"); + goto out; + } + cfg_info->nblocks = + SMS_HTONL(config_data[config_offset], + config_data[config_offset + 1], + config_data[config_offset + 2], config_data[config_offset + 3]); + config_offset += 4; + + cfg_info->blk_data = kcalloc( + cfg_info->nblocks, sizeof(struct tasdev_blk_data *), + GFP_KERNEL); + if (!cfg_info->blk_data) { + dev_err(tas_dev->dev, + "add config: blk_data alloc failed!\n"); + goto out; + } + cfg_info->real_nblocks = 0; + for (i = 0; i < (int)cfg_info->nblocks; i++) { + if (config_offset + 12 > config_size) { + dev_err(tas_dev->dev, + "add config: Out of memory: i = %d nblocks = %u!\n", + i, cfg_info->nblocks); + break; + } + cfg_info->blk_data[i] = kzalloc( + sizeof(struct tasdev_blk_data), GFP_KERNEL); + if (!cfg_info->blk_data[i]) + break; + + cfg_info->blk_data[i]->dev_idx = config_data[config_offset]; + config_offset++; + + cfg_info->blk_data[i]->block_type = config_data[config_offset]; + config_offset++; + + if (cfg_info->blk_data[i]->block_type == + TASDEVICE_BIN_BLK_PRE_POWER_UP) { + if (cfg_info->blk_data[i]->dev_idx == 0) { + cfg_info->active_dev = 1; + } else { + cfg_info->active_dev = + 1 << + (cfg_info->blk_data[i]->dev_idx - 1); + } + } + cfg_info->blk_data[i]->yram_checksum = + SMS_HTONS(config_data[config_offset], + config_data[config_offset + 1]); + config_offset += 2; + cfg_info->blk_data[i]->block_size = + SMS_HTONL(config_data[config_offset], + config_data[config_offset + 1], + config_data[config_offset + 2], + config_data[config_offset + 3]); + config_offset += 4; + + cfg_info->blk_data[i]->n_subblks = + SMS_HTONL(config_data[config_offset], + config_data[config_offset + 1], + config_data[config_offset + 2], + config_data[config_offset + 3]); + + config_offset += 4; + cfg_info->blk_data[i]->regdata = kzalloc( + cfg_info->blk_data[i]->block_size, GFP_KERNEL); + if (cfg_info->blk_data[i]->regdata == 0) { + dev_err(tas_dev->dev, + "add config: regdata alloc failed!\n"); + goto out; + } + if (config_offset + cfg_info->blk_data[i]->block_size + > config_size) { + dev_err(tas_dev->dev, + "%s: block_size Out of memory: i = %d nblocks = %u!\n", + __func__, i, + cfg_info->nblocks); + break; + } + memcpy(cfg_info->blk_data[i]->regdata, + &config_data[config_offset], + cfg_info->blk_data[i]->block_size); + config_offset += cfg_info->blk_data[i]->block_size; + cfg_info->real_nblocks += 1; + } +out: + return cfg_info; +} +static int tas2781_digital_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int val; + int ret = 0; + + /* Read the primary device as the whole */ + ret = tasdevice_dev_read(tas_dev, 0, mc->reg, &val); + if (ret) { + dev_err(tas_dev->dev, + "%s, get digital vol error\n", + __func__); + goto out; + } + val = (val > mc->max) ? mc->max : val; + val = mc->invert ? mc->max - val : val; + ucontrol->value.integer.value[0] = val; + +out: + return ret; +} + +static int tas2781_digital_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int val; + int i, ret = 0; + + val = ucontrol->value.integer.value[0]; + val = (val > mc->max) ? mc->max : val; + val = mc->invert ? mc->max - val : val; + val = (val < 0) ? 0 : val; + if (tas_dev->set_global_mode != NULL) { + ret = tasdevice_dev_write(tas_dev, tas_dev->ndev, + mc->reg, (unsigned int)val); + if (ret) + dev_err(tas_dev->dev, + "%s, set digital vol error in global mode\n", + __func__); + } else { + for (i = 0; i < tas_dev->ndev; i++) { + ret = tasdevice_dev_write(tas_dev, i, + mc->reg, (unsigned int)val); + if (ret) + dev_err(tas_dev->dev, + "%s, set digital vol error in device %d\n", + __func__, i); + } + } + + return ret; +} + +static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int val; + unsigned char mask = 0; + int ret = 0; + + /* Read the primary device */ + ret = tasdevice_dev_read(tas_dev, 0, mc->reg, &val); + if (ret) { + dev_err(tas_dev->dev, + "%s, get AMP vol error\n", + __func__); + goto out; + } + + mask = (1 << fls(mc->max)) - 1; + mask <<= mc->shift; + val = (val & mask) >> mc->shift; + val = (val > mc->max) ? mc->max : val; + val = mc->invert ? mc->max - val : val; + ucontrol->value.integer.value[0] = (int)val; + +out: + return ret; +} + +static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int val; + int i, ret = 0; + unsigned char mask = 0; + + mask = (1 << fls(mc->max)) - 1; + mask <<= mc->shift; + val = ucontrol->value.integer.value[0]; + val = (val > mc->max) ? mc->max : val; + val = mc->invert ? mc->max - val : val; + val = (val < 0) ? 0 : val; + for (i = 0; i < tas_dev->ndev; i++) { + ret = tasdevice_dev_update_bits(tas_dev, i, + mc->reg, mask, (unsigned int)(val << mc->shift)); + if (ret) + dev_err(tas_dev->dev, + "%s, set AMP vol error in device %d\n", + __func__, i); + } + + return ret; +} + +static const DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); +static const DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); + +static const struct snd_kcontrol_new tas2781_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Amp Gain Volume", TAS2781_AMP_LEVEL, + 1, 0, 20, 0, tas2781_amp_getvol, + tas2781_amp_putvol, amp_vol_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Digital Volume Control", TAS2781_DVC_LVL, + 0, 0, 200, 1, tas2781_digital_getvol, + tas2781_digital_putvol, dvc_tlv), +}; + +static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *p_tasdevice = + snd_soc_component_get_drvdata(codec); + + /* Codec Lock Hold*/ + mutex_lock(&p_tasdevice->codec_lock); + p_tasdevice->rcabin.profile_cfg_id = + ucontrol->value.integer.value[0]; + /* Codec Lock Release*/ + mutex_unlock(&p_tasdevice->codec_lock); + + return 0; +} + +static int tasdevice_info_programs(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *p_tasdevice = + snd_soc_component_get_drvdata(codec); + struct tasdevice_fw *tas_fw = p_tasdevice->fmw; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + /* Codec Lock Hold*/ + mutex_lock(&p_tasdevice->codec_lock); + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = (int)tas_fw->n_programs; + /* Codec Lock Release*/ + mutex_unlock(&p_tasdevice->codec_lock); + return 0; +} + +static int tasdevice_info_configurations( + struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *p_tasdevice = + snd_soc_component_get_drvdata(codec); + struct tasdevice_fw *tas_fw = p_tasdevice->fmw; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + /* Codec Lock Hold*/ + mutex_lock(&p_tasdevice->codec_lock); + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = (int)tas_fw->n_configurations - 1; + /* Codec Lock Release*/ + mutex_unlock(&p_tasdevice->codec_lock); + + return 0; +} + +static int tasdevice_info_profile(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *p_tasdevice = + snd_soc_component_get_drvdata(codec); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + /* Codec Lock Hold*/ + mutex_lock(&p_tasdevice->codec_lock); + uinfo->count = 1; + uinfo->value.integer.min = 1; + uinfo->value.integer.max = max(0, p_tasdevice->rcabin.ncfgs); + /* Codec Lock Release*/ + mutex_unlock(&p_tasdevice->codec_lock); + return 0; +} + +static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *p_tasdevice + = snd_soc_component_get_drvdata(codec); + + /* Codec Lock Hold*/ + mutex_lock(&p_tasdevice->codec_lock); + ucontrol->value.integer.value[0] = + p_tasdevice->rcabin.profile_cfg_id; + /* Codec Lock Release*/ + mutex_unlock(&p_tasdevice->codec_lock); + + return 0; +} + +static int tasdevice_create_controls(struct tasdevice_priv *tas_dev) +{ + int nr_controls = 1, ret = 0, mix_index = 0; + char *name = NULL; + struct snd_kcontrol_new *tasdevice_profile_controls = NULL; + + tasdevice_profile_controls = devm_kzalloc(tas_dev->dev, + nr_controls * sizeof(tasdevice_profile_controls[0]), + GFP_KERNEL); + if (tasdevice_profile_controls == NULL) { + ret = -ENOMEM; + goto out; + } + + /* Create a mixer item for selecting the active profile */ + name = devm_kzalloc(tas_dev->dev, + MAX_CONTROL_NAME, GFP_KERNEL); + if (!name) { + ret = -ENOMEM; + goto out; + } + scnprintf(name, MAX_CONTROL_NAME, "TASDEVICE Profile id"); + tasdevice_profile_controls[mix_index].name = name; + tasdevice_profile_controls[mix_index].iface = + SNDRV_CTL_ELEM_IFACE_MIXER; + tasdevice_profile_controls[mix_index].info = + tasdevice_info_profile; + tasdevice_profile_controls[mix_index].get = + tasdevice_get_profile_id; + tasdevice_profile_controls[mix_index].put = + tasdevice_set_profile_id; + mix_index++; + + ret = snd_soc_add_component_controls(tas_dev->codec, + tasdevice_profile_controls, + nr_controls < mix_index ? nr_controls : mix_index); + + tas_dev->tas_ctrl.nr_controls = + nr_controls < mix_index ? nr_controls : mix_index; +out: + return ret; +} + +static int tasdevice_program_get(struct snd_kcontrol *pKcontrol, + struct snd_ctl_elem_value *pValue) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(pKcontrol); + struct tasdevice_priv *pTAS2781 = snd_soc_component_get_drvdata(codec); + + mutex_lock(&pTAS2781->codec_lock); + pValue->value.integer.value[0] = pTAS2781->cur_prog; + mutex_unlock(&pTAS2781->codec_lock); + return 0; +} + +static int tasdevice_program_put(struct snd_kcontrol *pKcontrol, + struct snd_ctl_elem_value *pValue) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(pKcontrol); + struct tasdevice_priv *pTAS2781 = snd_soc_component_get_drvdata(codec); + unsigned int nProgram = pValue->value.integer.value[0]; + + mutex_lock(&pTAS2781->codec_lock); + pTAS2781->cur_prog = nProgram; + mutex_unlock(&pTAS2781->codec_lock); + return 0; +} + +static int tasdevice_configuration_get( + struct snd_kcontrol *pKcontrol, + struct snd_ctl_elem_value *pValue) +{ + + struct snd_soc_component *codec + = snd_soc_kcontrol_component(pKcontrol); + struct tasdevice_priv *pTAS2781 = snd_soc_component_get_drvdata(codec); + + mutex_lock(&pTAS2781->codec_lock); + pValue->value.integer.value[0] = pTAS2781->cur_conf; + mutex_unlock(&pTAS2781->codec_lock); + return 0; +} + +static int tasdevice_configuration_put( + struct snd_kcontrol *pKcontrol, + struct snd_ctl_elem_value *pValue) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(pKcontrol); + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(codec); + unsigned int n_configuration = pValue->value.integer.value[0]; + + mutex_lock(&tas_dev->codec_lock); + tas_dev->cur_conf = n_configuration; + mutex_unlock(&tas_dev->codec_lock); + return 0; +} + +static int tasdevice_dsp_create_control( + struct tasdevice_priv *tas_dev) +{ + int nr_controls = 2, ret = 0, mix_index = 0; + char *program_name = NULL; + char *configuration_name = NULL; + struct snd_kcontrol_new *tasdevice_dsp_controls = NULL; + + tasdevice_dsp_controls = devm_kzalloc(tas_dev->dev, + nr_controls * sizeof(tasdevice_dsp_controls[0]), + GFP_KERNEL); + if (tasdevice_dsp_controls == NULL) { + ret = -ENOMEM; + goto out; + } + + /* Create a mixer item for selecting the active profile */ + program_name = devm_kzalloc(tas_dev->dev, + MAX_CONTROL_NAME, GFP_KERNEL); + configuration_name = devm_kzalloc(tas_dev->dev, + MAX_CONTROL_NAME, GFP_KERNEL); + if (!program_name || !configuration_name) { + ret = -ENOMEM; + goto out; + } + + scnprintf(program_name, MAX_CONTROL_NAME, "Program"); + tasdevice_dsp_controls[mix_index].name = program_name; + tasdevice_dsp_controls[mix_index].iface = + SNDRV_CTL_ELEM_IFACE_MIXER; + tasdevice_dsp_controls[mix_index].info = + tasdevice_info_programs; + tasdevice_dsp_controls[mix_index].get = + tasdevice_program_get; + tasdevice_dsp_controls[mix_index].put = + tasdevice_program_put; + mix_index++; + + scnprintf(configuration_name, MAX_CONTROL_NAME, "Configuration"); + tasdevice_dsp_controls[mix_index].name = configuration_name; + tasdevice_dsp_controls[mix_index].iface = + SNDRV_CTL_ELEM_IFACE_MIXER; + tasdevice_dsp_controls[mix_index].info = + tasdevice_info_configurations; + tasdevice_dsp_controls[mix_index].get = + tasdevice_configuration_get; + tasdevice_dsp_controls[mix_index].put = + tasdevice_configuration_put; + mix_index++; + + ret = snd_soc_add_component_controls(tas_dev->codec, + tasdevice_dsp_controls, + nr_controls < mix_index ? nr_controls : mix_index); + + tas_dev->tas_ctrl.nr_controls += nr_controls; +out: + return ret; +} + +static void tasdevice_rca_ready(const struct firmware *fmw, + void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + struct tasdevice_rca *rca = NULL; + struct tasdevice_rca_hdr *fw_hdr = NULL; + struct tasdevice_config_info **cfg_info = NULL; + const struct firmware *fw_entry = NULL; + unsigned char *buf = NULL; + int offset = 0, i = 0; + unsigned int total_config_sz = 0; + int ret = 0; + + mutex_lock(&tas_dev->codec_lock); + rca = &(tas_dev->rcabin); + fw_hdr = &(rca->fw_hdr); + if (unlikely(!fmw) || unlikely(!fmw->data)) { + dev_err(tas_dev->dev, + "Failed to read %s, no side - effect on driver running\n", + tas_dev->rca_binaryname); + ret = -1; + goto out; + } + buf = (unsigned char *)fmw->data; + + dev_info(tas_dev->dev, "tasdev: rca_ready start\n"); + fw_hdr->img_sz = SMS_HTONL(buf[offset], buf[offset + 1], + buf[offset + 2], buf[offset + 3]); + offset += 4; + if (fw_hdr->img_sz != fmw->size) { + dev_err(tas_dev->dev, + "File size not match, %d %u", (int)fmw->size, + fw_hdr->img_sz); + ret = -1; + goto out; + } + + fw_hdr->checksum = SMS_HTONL(buf[offset], buf[offset + 1], + buf[offset + 2], buf[offset + 3]); + offset += 4; + fw_hdr->binary_version_num = SMS_HTONL(buf[offset], + buf[offset + 1], buf[offset + 2], buf[offset + 3]); + if (fw_hdr->binary_version_num < 0x103) { + dev_err(tas_dev->dev, + "File version 0x%04x is too low", + fw_hdr->binary_version_num); + ret = -1; + goto out; + } + offset += 4; + fw_hdr->drv_fw_version = SMS_HTONL(buf[offset], buf[offset + 1], + buf[offset + 2], buf[offset + 3]); + offset += 8; + fw_hdr->plat_type = buf[offset]; + offset += 1; + fw_hdr->dev_family = buf[offset]; + offset += 1; + fw_hdr->reserve = buf[offset]; + offset += 1; + fw_hdr->ndev = buf[offset]; + offset += 1; + if (fw_hdr->ndev != tas_dev->ndev) { + dev_err(tas_dev->dev, + "ndev(%u) from rcabin and ndev(%u) from DTS does not match\n", + fw_hdr->ndev, + tas_dev->ndev); + ret = -1; + goto out; + } + if (offset + TASDEVICE_DEVICE_SUM > fw_hdr->img_sz) { + dev_err(tas_dev->dev, + "rca_ready: Out of Memory!\n"); + ret = -1; + goto out; + } + + for (i = 0; i < TASDEVICE_DEVICE_SUM; i++, offset++) + fw_hdr->devs[i] = buf[offset]; + + fw_hdr->nconfig = SMS_HTONL(buf[offset], buf[offset + 1], + buf[offset + 2], buf[offset + 3]); + offset += 4; + dev_info(tas_dev->dev, "nconfig = %u\n", fw_hdr->nconfig); + for (i = 0; i < TASDEVICE_CONFIG_SUM; i++) { + fw_hdr->config_size[i] = SMS_HTONL(buf[offset], + buf[offset + 1], buf[offset + 2], buf[offset + 3]); + offset += 4; + total_config_sz += fw_hdr->config_size[i]; + } + dev_info(tas_dev->dev, + "img_sz = %u total_config_sz = %u offset = %d\n", + fw_hdr->img_sz, total_config_sz, offset); + if (fw_hdr->img_sz - total_config_sz != (unsigned int)offset) { + dev_err(tas_dev->dev, "Bin file error!\n"); + ret = -1; + goto out; + } + cfg_info = kcalloc(fw_hdr->nconfig, + sizeof(struct tasdevice_config_info *), + GFP_KERNEL); + + if (!cfg_info) { + ret = -1; + dev_err(tas_dev->dev, "nconfig Memory alloc failed!\n"); + goto out; + } + rca->cfg_info = cfg_info; + rca->ncfgs = 0; + for (i = 0; i < (int)fw_hdr->nconfig; i++) { + cfg_info[i] = tasdevice_add_config(context, &buf[offset], + fw_hdr->config_size[i]); + if (!cfg_info[i]) { + ret = -1; + dev_err(tas_dev->dev, + "add_config Memory alloc failed!\n"); + break; + } + offset += (int)fw_hdr->config_size[i]; + rca->ncfgs += 1; + } + tasdevice_create_controls(tas_dev); + + tasdevice_dsp_remove(tas_dev); + tasdevice_calbin_remove(tas_dev); + tas_dev->fw_state = TASDEVICE_DSP_FW_PENDING; + scnprintf(tas_dev->coef_binaryname, 64, "%s_coef.bin", + tas_dev->dev_name); + ret = request_firmware(&fw_entry, tas_dev->coef_binaryname, + tas_dev->dev); + if (!ret) { + ret = tasdevice_dspfw_ready(fw_entry, tas_dev); + release_firmware(fw_entry); + fw_entry = NULL; + } else { + tas_dev->fw_state = TASDEVICE_DSP_FW_FAIL; + dev_err(tas_dev->dev, "%s: load %s error\n", __func__, + tas_dev->coef_binaryname); + goto out; + } + tasdevice_dsp_create_control(tas_dev); + + tas_dev->fw_state = TASDEVICE_DSP_FW_ALL_OK; + tas_dev->is_glb_calibrated_data_loaded = true; + for (i = 0; i < tas_dev->ndev; i++) { + scnprintf(tas_dev->cal_binaryname[i], 64, "%s_cal_0x%02x.bin", + tas_dev->dev_name, tas_dev->tasdevice[i].dev_addr); + ret = tas2781_load_calibration(tas_dev, + tas_dev->cal_binaryname[i], i); + if (ret != 0) { + dev_err(tas_dev->dev, + "%s: load %s error, no-side effect\n", + __func__, + tas_dev->cal_binaryname[i]); + ret = 0; + tas_dev->is_glb_calibrated_data_loaded = false; + } + } + +out: + mutex_unlock(&tas_dev->codec_lock); + if (fmw) + release_firmware(fmw); + dev_info(tas_dev->dev, "Firmware init complete\n"); +} + +static void tasdevice_config_info_remove(void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + struct tasdevice_rca *rca = &(tas_dev->rcabin); + struct tasdevice_config_info **cfg_info = rca->cfg_info; + int i = 0, j = 0; + + mutex_lock(&tas_dev->dev_lock); + if (cfg_info) { + for (i = 0; i < rca->ncfgs; i++) { + if (cfg_info[i]) { + for (j = 0; j < (int)cfg_info[i]->real_nblocks; + j++) { + kfree( + cfg_info[i]->blk_data[j]->regdata); + kfree(cfg_info[i]->blk_data[j]); + } + kfree(cfg_info[i]->blk_data); + kfree(cfg_info[i]); + } + } + kfree(cfg_info); + } + mutex_unlock(&tas_dev->dev_lock); +} + +static void tasdevice_enable_irq( + struct tasdevice_priv *tas_dev, bool enable) +{ + if (enable) { + if (tas_dev->irq_info.is_irq_enable) + return; + if (gpio_is_valid(tas_dev->irq_info.irq_gpio)) + enable_irq(tas_dev->irq_info.irq); + tas_dev->irq_info.is_irq_enable = true; + } else { + if (gpio_is_valid(tas_dev->irq_info.irq_gpio)) + disable_irq_nosync(tas_dev->irq_info.irq); + tas_dev->irq_info.is_irq_enable = false; + } +} + +static void tasdevice_set_power_state( + struct tasdevice_priv *tas_dev, int state) +{ + switch (state) { + case 0: + schedule_delayed_work(&tas_dev->tuning_switch_wrk, + msecs_to_jiffies(20)); + break; + default: + if (!(tas_dev->pstream || tas_dev->cstream)) { + if (tas_dev->irq_work_func) + tasdevice_enable_irq(tas_dev, false); + tasdevice_select_cfg_blk(tas_dev, + tas_dev->rcabin.profile_cfg_id, + TASDEVICE_BIN_BLK_PRE_SHUTDOWN); + } + break; + } +} + +static int tasdevice_dapm_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *codec = snd_soc_dapm_to_component(w->dapm); + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(codec); + int stream; + + /* Codec Lock Hold */ + mutex_lock(&tas_dev->codec_lock); + stream = strcmp(w->name, "SPK") ? SNDRV_PCM_STREAM_PLAYBACK : + SNDRV_PCM_STREAM_CAPTURE; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + dev_info(tas_dev->dev, "SND_SOC_DAPM_POST_PMU\n"); + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + tas_dev->pstream = 1; + else + tas_dev->cstream = 1; + tasdevice_set_power_state(tas_dev, 0); + break; + case SND_SOC_DAPM_PRE_PMD: + dev_info(tas_dev->dev, "SND_SOC_DAPM_PRE_PMD\n"); + /* stop DSP only when both playback and capture streams + * are deactivated + */ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + tas_dev->pstream = 0; + else + tas_dev->cstream = 0; + if (tas_dev->pstream != 0 || tas_dev->cstream != 0) + goto out; + tasdevice_set_power_state(tas_dev, 1); + break; + } + /* Codec Lock Release*/ + mutex_unlock(&tas_dev->codec_lock); +out: + return 0; +} + +static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT_E("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, + 0, 0, tasdevice_dapm_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SPK("SPK", tasdevice_dapm_event), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_INPUT("DMIC") +}; + +static const struct snd_soc_dapm_route tasdevice_audio_map[] = { + {"SPK", NULL, "ASI"}, + {"OUT", NULL, "SPK"}, + {"ASI OUT", NULL, "DMIC"} +}; + +static int tasdevice_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *codec = dai->component; + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(codec); + int ret = 0; + + if (tas_dev->fw_state != TASDEVICE_DSP_FW_ALL_OK) { + dev_err(tas_dev->dev, "DSP bin file not loaded\n"); + ret = -EINVAL; + } + return ret; +} + +static int tasdevice_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct tasdevice_priv *tas_dev = snd_soc_dai_get_drvdata(dai); + unsigned int fsrate; + unsigned int slot_width; + int bclk_rate; + int rc = 0; + + dev_info(tas_dev->dev, "%s: %s\n", + __func__, substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + "Playback":"Capture"); + + fsrate = params_rate(params); + switch (fsrate) { + case 48000: + break; + case 44100: + break; + default: + dev_err(tas_dev->dev, + "%s: incorrect sample rate = %u\n", + __func__, fsrate); + rc = -EINVAL; + goto out; + } + + slot_width = params_width(params); + switch (slot_width) { + case 16: + break; + case 20: + break; + case 24: + break; + case 32: + break; + default: + dev_err(tas_dev->dev, + "%s: incorrect slot width = %u\n", + __func__, slot_width); + rc = -EINVAL; + goto out; + } + + bclk_rate = snd_soc_params_to_bclk(params); + if (bclk_rate < 0) { + dev_err(tas_dev->dev, + "%s: incorrect bclk rate = %d\n", + __func__, bclk_rate); + rc = bclk_rate; + goto out; + } + dev_info(tas_dev->dev, + "%s: BCLK rate = %d Channel = %d Sample rate = %u slot width = %u\n", + __func__, bclk_rate, params_channels(params), + fsrate, slot_width); +out: + return rc; +} + +static int tasdevice_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct tasdevice_priv *tas_dev = snd_soc_dai_get_drvdata(codec_dai); + + dev_info(tas_dev->dev, + "%s: clk_id = %d, freq = %u, CLK direction %s\n", + __func__, clk_id, freq, + dir == TASDEVICE_CLK_DIR_OUT ? "OUT":"IN"); + + return 0; +} + +static void tuning_switch_routine(struct work_struct *work) +{ + struct tasdevice_priv *tas_dev = + container_of(work, struct tasdevice_priv, + tuning_switch_wrk.work); + struct tasdevice_fw *tas_fmw = tas_dev->fmw; + int profile_cfg_id = 0; + int is_set_glb_mode = 0; + + mutex_lock(&tas_dev->codec_lock); + + if (tas_fmw) { + if (tas_dev->cur_prog >= tas_fmw->n_programs) + /*bypass all in rca is profile id 0*/ + profile_cfg_id = RCA_CONFIGID_BYPASS_ALL; + else { + /*dsp mode or tuning mode*/ + profile_cfg_id = tas_dev->rcabin.profile_cfg_id; + tas_fmw = tas_dev->fmw; + dev_info(tas_dev->dev, "%s: %s\n", __func__, + tas_fmw->configs[tas_dev->cur_conf].name); + is_set_glb_mode = + tasdevice_select_tuningprm_cfg(tas_dev, + tas_dev->cur_prog, + tas_dev->cur_conf, + profile_cfg_id); + if (tas_dev->set_global_mode != NULL) + tas_dev->set_global_mode(tas_dev); + } + } else + profile_cfg_id = RCA_CONFIGID_BYPASS_ALL; + + tasdevice_select_cfg_blk(tas_dev, profile_cfg_id, + TASDEVICE_BIN_BLK_PRE_POWER_UP); + + if (is_set_glb_mode && tas_dev->set_global_mode) + tas_dev->set_global_mode(tas_dev); + if (tas_dev->irq_work_func) + tasdevice_enable_irq(tas_dev, true); + mutex_unlock(&tas_dev->codec_lock); +} + +static const struct snd_soc_dai_ops tasdevice_dai_ops = { + .startup = tasdevice_startup, + .hw_params = tasdevice_hw_params, + .set_sysclk = tasdevice_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver tasdevice_dai_driver[] = { + { + .name = "tas2781_codec", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TASDEVICE_RATES, + .formats = TASDEVICE_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = TASDEVICE_RATES, + .formats = TASDEVICE_FORMATS, + }, + .ops = &tasdevice_dai_ops, + .symmetric_rate = 1, + }, +}; + +static void tas2781_reset(struct tasdevice_priv *tas_dev) +{ + int ret = 0; + int i = 0; + + for (; i < tas_dev->ndev; i++) { + ret = tasdevice_dev_write(tas_dev, i, + TAS2871_REG_SWRESET, + TAS2871_REG_SWRESET_RESET); + if (ret < 0) { + dev_err(tas_dev->dev, "%s: chn %d reset fail, %d\n", + __func__, i, ret); + continue; + } + } + usleep_range(1000, 1050); +} + +static int tasdevice_codec_probe( + struct snd_soc_component *codec) +{ + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + int ret; + + /* Codec Lock Hold */ + mutex_lock(&tas_dev->codec_lock); + + tas_dev->codec = codec; + scnprintf(tas_dev->rca_binaryname, 64, "%s_rca.bin", + tas_dev->dev_name); + ret = request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, + tas_dev->rca_binaryname, tas_dev->dev, GFP_KERNEL, tas_dev, + tasdevice_rca_ready); + if (ret) + dev_err(tas_dev->dev, + "%s: request_firmware_nowait error:0x%08x\n", + __func__, ret); + + /* Codec Lock Release*/ + mutex_unlock(&tas_dev->codec_lock); + + tas2781_reset(tas_dev); + if (tas_dev->set_global_mode != NULL) + tas_dev->set_global_mode(tas_dev); + + return ret; +} + +static void tasdevice_deinit(void *context) +{ + struct tasdevice_priv *tas_dev = (struct tasdevice_priv *) context; + + tasdevice_config_info_remove(tas_dev); + tasdevice_dsp_remove(tas_dev); + tasdevice_calbin_remove(tas_dev); + tas_dev->fw_state = TASDEVICE_DSP_FW_PENDING; +} + +static void tasdevice_codec_remove( + struct snd_soc_component *codec) +{ + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + /* Codec Lock Hold */ + mutex_lock(&tas_dev->codec_lock); + tasdevice_deinit(tas_dev); + /* Codec Lock Release*/ + mutex_unlock(&tas_dev->codec_lock); +} + +static const struct snd_soc_component_driver + soc_codec_driver_tasdevice = { + .probe = tasdevice_codec_probe, + .remove = tasdevice_codec_remove, + .controls = tas2781_snd_controls, + .num_controls = ARRAY_SIZE(tas2781_snd_controls), + .dapm_widgets = tasdevice_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), + .dapm_routes = tasdevice_audio_map, + .num_dapm_routes = ARRAY_SIZE(tasdevice_audio_map), + .idle_bias_on = 1, + .endianness = 1, +}; + +static void irq_work_routine(struct work_struct *pWork) +{ + struct tasdevice_priv *tas_dev = + container_of(pWork, struct tasdevice_priv, + irq_info.irq_work.work); + + mutex_lock(&tas_dev->codec_lock); + if (tas_dev->runtime_suspend) { + dev_info(tas_dev->dev, "%s, Runtime Suspended\n", __func__); + goto end; + } + /*Logical Layer IRQ function, return is ignored*/ + if (tas_dev->irq_work_func) + tas_dev->irq_work_func(tas_dev); + else + dev_info(tas_dev->dev, + "%s, irq_work_func is NULL\n", __func__); +end: + mutex_unlock(&tas_dev->codec_lock); +} + +static void tas2781_irq_work_func(struct tasdevice_priv *tas_dev) +{ + int rc = 0; + unsigned int reg_val = 0, array_size = 0, i = 0, ndev = 0; + unsigned int int_reg_array[] = { + TAS2781_REG_INT_LTCH0, + TAS2781_REG_INT_LTCH1, + TAS2781_REG_INT_LTCH1_0, + TAS2781_REG_INT_LTCH2, + TAS2781_REG_INT_LTCH3, + TAS2781_REG_INT_LTCH4}; + + tasdevice_enable_irq(tas_dev, false); + + array_size = ARRAY_SIZE(int_reg_array); + + for (ndev = 0; ndev < tas_dev->ndev; ndev++) { + for (i = 0; i < array_size; i++) { + rc = tasdevice_dev_read(tas_dev, + ndev, int_reg_array[i], &reg_val); + if (!rc) + dev_info(tas_dev->dev, + "INT STATUS REG 0x%04x=0x%02x\n", + int_reg_array[i], reg_val); + else + dev_err(tas_dev->dev, + "Read Reg 0x%04x error(rc=%d)\n", + int_reg_array[i], rc); + } + } + +} + +static irqreturn_t tasdevice_irq_handler(int irq, + void *dev_id) +{ + struct tasdevice_priv *tas_dev = (struct tasdevice_priv *)dev_id; + + /* get IRQ status after 100 ms */ + schedule_delayed_work(&tas_dev->irq_info.irq_work, + msecs_to_jiffies(100)); + return IRQ_HANDLED; +} + +static int tasdevice_parse_dt(struct tasdevice_priv *tas_dev) +{ + struct device_node *np = tas_dev->dev->of_node; + int rc = 0, i = 0, ndev = 0; + unsigned int dev_addrs[max_chn]; + struct i2c_client *client = (struct i2c_client *)tas_dev->client; + + ndev = of_property_read_variable_u32_array(np, "ti,audio-slots", + dev_addrs, 0, ARRAY_SIZE(dev_addrs)); + if (ndev <= 0) { + ndev = 1; + dev_addrs[0] = client->addr; + } + for (i = 0; i < ndev; i++) { + tas_dev->tasdevice[i].dev_addr = dev_addrs[i]; + dev_info(tas_dev->dev, "%s=0x%02x", + dts_dev_addr_tag[i], + tas_dev->tasdevice[i].dev_addr); + } + if (ndev > 1) { + rc = of_property_read_u32(np, "ti,broadcast-addr", + &(tas_dev->glb_addr.dev_addr)); + if (rc) { + dev_err(tas_dev->dev, + "Looking up node %s failed %d\n", + np->full_name, rc); + tas_dev->glb_addr.dev_addr = 0; + } + } + + tas_dev->ndev = ndev; + + tas_dev->rst_gpio = of_get_named_gpio(np, "reset-gpios", i); + if (gpio_is_valid(tas_dev->rst_gpio)) { + rc = gpio_request(tas_dev->rst_gpio, "reset"); + if (!rc) { + gpio_direction_output(tas_dev->rst_gpio, 1); + dev_info(tas_dev->dev, "reset-gpio = %d", + tas_dev->rst_gpio); + ndev++; + } else + dev_err(tas_dev->dev, + "%s: Failed to request gpio %d\n", + __func__, tas_dev->rst_gpio); + } else + dev_err(tas_dev->dev, + "Looking up node %s failed %d\n", + np->full_name, + tas_dev->rst_gpio); + + strcpy(tas_dev->dev_name, tasdevice_id[tas_dev->chip_id].name); + + tas_dev->irq_info.irq_gpio = of_get_named_gpio(np, + "interrupts", 0); + if (gpio_is_valid(tas_dev->irq_info.irq_gpio)) { + dev_dbg(tas_dev->dev, "interrupts gpio = %d", + tas_dev->irq_info.irq_gpio); + INIT_DELAYED_WORK(&tas_dev->irq_info.irq_work, + irq_work_routine); + + rc = gpio_request(tas_dev->irq_info.irq_gpio, + "AUDEV-IRQ"); + if (!rc) { + gpio_direction_input( + tas_dev->irq_info.irq_gpio); + + tas_dev->irq_info.irq = + gpio_to_irq( + tas_dev->irq_info.irq_gpio); + dev_info(tas_dev->dev, + "irq = %d\n", + tas_dev->irq_info.irq); + + rc = request_threaded_irq( + tas_dev->irq_info.irq, + tasdevice_irq_handler, + NULL, IRQF_TRIGGER_FALLING| + IRQF_ONESHOT, + SMARTAMP_MODULE_NAME, tas_dev); + if (!rc) + disable_irq_nosync( + tas_dev->irq_info.irq); + else + dev_err(tas_dev->dev, + "request_irq failed, %d\n", + rc); + } else + dev_err(tas_dev->dev, + "%s: GPIO %d request error\n", + __func__, + tas_dev->irq_info.irq_gpio); + } else + dev_err(tas_dev->dev, + "Looking up irq-gpio property in node %s failed %d\n", + np->full_name, + tas_dev->irq_info.irq_gpio); + + if (gpio_is_valid(tas_dev->irq_info.irq_gpio) && rc == 0) { + if (tas_dev->chip_id == TAS2781) + tas_dev->irq_work_func = tas2781_irq_work_func; + else + dev_info(tas_dev->dev, "%s: No match irq_work_func\n", + __func__); + } + + return 0; +} + +static int tasdevice_change_chn_book(struct tasdevice_priv *tas_dev, + enum channel chn, int book) +{ + int n_result = 0; + struct i2c_client *pClient = + (struct i2c_client *)tas_dev->client; + + if (chn < tas_dev->ndev) { + if (tas_dev->glb_addr.ref_cnt != 0) { + tas_dev->glb_addr.ref_cnt = 0; + tas_dev->glb_addr.cur_book = -1; + } + pClient->addr = tas_dev->tasdevice[chn].dev_addr; + if (tas_dev->tasdevice[chn].cur_book != book) { + n_result = regmap_write(tas_dev->regmap, + TASDEVICE_BOOKCTL_PAGE, 0); + if (n_result < 0) { + dev_err(tas_dev->dev, "%s, ERROR, E=%d\n", + __func__, n_result); + goto out; + } + n_result = regmap_write(tas_dev->regmap, + TASDEVICE_BOOKCTL_REG, book); + if (n_result < 0) { + dev_err(tas_dev->dev, "%s, ERROR, E=%d\n", + __func__, n_result); + goto out; + } + tas_dev->tasdevice[chn].cur_book = book; + } + } else if (chn == tas_dev->ndev) { + int i = 0; + + if (tas_dev->glb_addr.ref_cnt == 0) + for (i = 0; i < tas_dev->ndev; i++) + tas_dev->tasdevice[i].cur_book = -1; + pClient->addr = tas_dev->glb_addr.dev_addr; + if (tas_dev->glb_addr.cur_book != book) { + n_result = regmap_write(tas_dev->regmap, + TASDEVICE_BOOKCTL_PAGE, 0); + if (n_result < 0) { + dev_err(tas_dev->dev, + "%s, 0ERROR, E=%d\n", + __func__, n_result); + goto out; + } + n_result = regmap_write(tas_dev->regmap, + TASDEVICE_BOOKCTL_REG, book); + if (n_result < 0) { + dev_err(tas_dev->dev, + "%s, book%xERROR, E=%d\n", + __func__, book, n_result); + goto out; + } + tas_dev->glb_addr.cur_book = book; + } + + tas_dev->glb_addr.ref_cnt++; + } else + dev_err(tas_dev->dev, + "%s, ERROR, no such channel(%d)\n", + __func__, chn); + +out: + return n_result; +} + +int tasdevice_dev_read(struct tasdevice_priv *tas_dev, + enum channel chn, unsigned int reg, unsigned int *pValue) +{ + int n_result = 0; + + mutex_lock(&tas_dev->dev_lock); + if (chn < tas_dev->ndev) { + n_result = tasdevice_change_chn_book(tas_dev, chn, + TASDEVICE_BOOK_ID(reg)); + if (n_result < 0) + goto out; + + n_result = regmap_read(tas_dev->regmap, + TASDEVICE_PGRG(reg), pValue); + if (n_result < 0) + dev_err(tas_dev->dev, "%s, ERROR,E=%d\n", + __func__, n_result); + } else { + + dev_err(tas_dev->dev, "%s, ERROR, no such channel(%d)\n", + __func__, chn); + } + +out: + mutex_unlock(&tas_dev->dev_lock); + return n_result; +} + +int tasdevice_dev_write(struct tasdevice_priv *tas_dev, + enum channel chn, unsigned int reg, unsigned int value) +{ + int n_result = 0; + + mutex_lock(&tas_dev->dev_lock); + if (chn <= tas_dev->ndev) { + n_result = tasdevice_change_chn_book(tas_dev, chn, + TASDEVICE_BOOK_ID(reg)); + if (n_result < 0) + goto out; + + n_result = regmap_write(tas_dev->regmap, + TASDEVICE_PGRG(reg), value); + if (n_result < 0) + dev_err(tas_dev->dev, "%s, ERROR, E=%d\n", + __func__, n_result); + } else + dev_err(tas_dev->dev, "%s, ERROR, no such channel(%d)\n", + __func__, chn); +out: + mutex_unlock(&tas_dev->dev_lock); + return n_result; +} + +int tasdevice_dev_bulk_write( + struct tasdevice_priv *tas_dev, enum channel chn, + unsigned int reg, unsigned char *p_data, + unsigned int n_length) +{ + int n_result = 0; + + mutex_lock(&tas_dev->dev_lock); + if (chn <= tas_dev->ndev) { + n_result = tasdevice_change_chn_book(tas_dev, chn, + TASDEVICE_BOOK_ID(reg)); + if (n_result < 0) + goto out; + + n_result = regmap_bulk_write(tas_dev->regmap, + TASDEVICE_PGRG(reg), p_data, n_length); + if (n_result < 0) + dev_err(tas_dev->dev, "%s, ERROR, E=%d\n", + __func__, n_result); + } else + dev_err(tas_dev->dev, "%s, ERROR, no such channel(%d)\n", + __func__, chn); +out: + mutex_unlock(&tas_dev->dev_lock); + return n_result; +} + +int tasdevice_dev_bulk_read(struct tasdevice_priv *tas_dev, + enum channel chn, unsigned int reg, unsigned char *p_data, + unsigned int n_length) +{ + int n_result = 0; + + mutex_lock(&tas_dev->dev_lock); + if (chn < tas_dev->ndev) { + n_result = tasdevice_change_chn_book(tas_dev, chn, + TASDEVICE_BOOK_ID(reg)); + if (n_result < 0) + goto out; + + n_result = regmap_bulk_read(tas_dev->regmap, + TASDEVICE_PGRG(reg), p_data, n_length); + if (n_result < 0) + dev_err(tas_dev->dev, "%s, ERROR, E=%d\n", + __func__, n_result); + } else + dev_err(tas_dev->dev, "%s, ERROR, no such channel(%d)\n", + __func__, chn); + +out: + mutex_unlock(&tas_dev->dev_lock); + return n_result; +} + +int tasdevice_dev_update_bits( + struct tasdevice_priv *tas_dev, enum channel chn, + unsigned int reg, unsigned int mask, unsigned int value) +{ + int n_result = 0; + struct i2c_client *pClient = + (struct i2c_client *)tas_dev->client; + + mutex_lock(&tas_dev->dev_lock); + if (chn < tas_dev->ndev) { + n_result = tasdevice_change_chn_book(tas_dev, chn, + TASDEVICE_BOOK_ID(reg)); + if (n_result < 0) + goto out; + pClient->addr = tas_dev->tasdevice[chn].dev_addr; + n_result = regmap_update_bits(tas_dev->regmap, + TASDEVICE_PGRG(reg), mask, value); + if (n_result < 0) + dev_err(tas_dev->dev, "%s, ERROR, E=%d\n", + __func__, n_result); + } else { + dev_err(tas_dev->dev, "%s, ERROR, no such channel(%d)\n", + __func__, chn); + n_result = -1; + } + +out: + mutex_unlock(&tas_dev->dev_lock); + return n_result; +} + +static void tas2781_set_global_mode(struct tasdevice_priv *tas_dev) +{ + int i = 0; + int ret = 0; + + for (; i < tas_dev->ndev; i++) { + ret = tasdevice_dev_update_bits(tas_dev, i, + TAS2871_MISC_CFG2, TAS2871_GLOBAL_ADDR_MASK, + TAS2871_GLOBAL_ADDR_ENABLE); + if (ret < 0) { + dev_err(tas_dev->dev, + "%s: chn %d set global fail, %d\n", + __func__, i, ret); + continue; + } + } +} + +static int tasdevice_init(struct tasdevice_priv *tas_dev) +{ + int ret = 0, i = 0; + + tas_dev->cur_prog = -1; + tas_dev->cur_conf = -1; + + for (i = 0; i < tas_dev->ndev; i++) { + tas_dev->tasdevice[i].cur_book = -1; + tas_dev->tasdevice[i].cur_prog = -1; + tas_dev->tasdevice[i].cur_conf = -1; + } + mutex_init(&tas_dev->dev_lock); + if (tas_dev->glb_addr.dev_addr != 0 + && tas_dev->glb_addr.dev_addr < 0x7F) + tas_dev->set_global_mode = tas2781_set_global_mode; + dev_set_drvdata(tas_dev->dev, tas_dev); + + INIT_DELAYED_WORK(&tas_dev->tuning_switch_wrk, + tuning_switch_routine); + + mutex_init(&tas_dev->codec_lock); + ret = devm_snd_soc_register_component(tas_dev->dev, + &soc_codec_driver_tasdevice, + tasdevice_dai_driver, ARRAY_SIZE(tasdevice_dai_driver)); + if (ret) + dev_err(tas_dev->dev, "%s: codec register error:0x%08x\n", + __func__, ret); + + INIT_DELAYED_WORK(&tas_dev->irq_info.irq_work, irq_work_routine); + tas_dev->irq_info.is_irq_enable = false; + + dev_info(tas_dev->dev, "i2c register success\n"); + + return ret; +} + +static void tasdevice_remove(struct tasdevice_priv *tas_dev) +{ + if (gpio_is_valid(tas_dev->rst_gpio)) + gpio_free(tas_dev->rst_gpio); + + if (gpio_is_valid(tas_dev->irq_info.irq_gpio)) { + if (delayed_work_pending(&tas_dev->irq_info.irq_work)) { + dev_info(tas_dev->dev, "cancel IRQ work\n"); + cancel_delayed_work(&tas_dev->irq_info.irq_work); + } + cancel_delayed_work_sync(&tas_dev->irq_info.irq_work); + } + + mutex_destroy(&tas_dev->dev_lock); + mutex_destroy(&tas_dev->codec_lock); + snd_soc_unregister_component(tas_dev->dev); +} + +static int tasdevice_pm_suspend(struct device *dev) +{ + struct tasdevice_priv *tas_dev = dev_get_drvdata(dev); + + mutex_lock(&tas_dev->codec_lock); + tas_dev->runtime_suspend = true; + + if (tas_dev->irq_work_func) { + if (delayed_work_pending(&tas_dev->irq_info.irq_work)) { + dev_dbg(tas_dev->dev, "cancel IRQ work\n"); + cancel_delayed_work_sync(&tas_dev->irq_info.irq_work); + } + } + mutex_unlock(&tas_dev->codec_lock); + return 0; +} + +static int tasdevice_pm_resume(struct device *dev) +{ + struct tasdevice_priv *tas_dev = dev_get_drvdata(dev); + + mutex_lock(&tas_dev->codec_lock); + tas_dev->runtime_suspend = false; + mutex_unlock(&tas_dev->codec_lock); + return 0; +} + +const struct dev_pm_ops tasdevice_pm_ops = { + .suspend = tasdevice_pm_suspend, + .resume = tasdevice_pm_resume +}; + +static int tasdevice_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct tasdevice_priv *tas_dev = NULL; + int ret = 0; + + if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_I2C)) { + dev_err(&i2c->dev, + "%s: I2C check failed\n", __func__); + ret = -ENODEV; + goto out; + } + + tas_dev = devm_kzalloc(&i2c->dev, sizeof(*tas_dev), GFP_KERNEL); + if (!tas_dev) { + ret = -ENOMEM; + goto out; + } + tas_dev->dev = &i2c->dev; + tas_dev->client = (void *)i2c; + tas_dev->chip_id = id->driver_data; + + if (i2c->dev.of_node) + ret = tasdevice_parse_dt(tas_dev); + else { + dev_err(tas_dev->dev, "No DTS info\n"); + goto out; + } + + tas_dev->regmap = devm_regmap_init_i2c(i2c, + &tasdevice_regmap); + if (IS_ERR(tas_dev->regmap)) { + ret = PTR_ERR(tas_dev->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + ret = tasdevice_init(tas_dev); + +out: + if (ret < 0 && tas_dev != NULL) + tasdevice_remove(tas_dev); + return ret; + +} + +static void tasdevice_i2c_remove(struct i2c_client *pClient) +{ + struct tasdevice_priv *tas_dev = i2c_get_clientdata(pClient); + + if (tas_dev) + tasdevice_remove(tas_dev); + +} + +static struct i2c_driver tasdevice_i2c_driver = { + .driver = { + .name = "tas2781-codec", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tasdevice_of_match), + .pm = &tasdevice_pm_ops, + }, + .probe = tasdevice_i2c_probe, + .remove = tasdevice_i2c_remove, + .id_table = tasdevice_id, +}; + +module_i2c_driver(tasdevice_i2c_driver); + +MODULE_AUTHOR("Shenghao Ding <shenghao-ding(a)ti.com>"); +MODULE_AUTHOR("Kevin Lu <kevin-lu(a)ti.com>"); +MODULE_DESCRIPTION("ASoC TAS2781 Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2781.h b/sound/soc/codecs/tas2781.h new file mode 100644 index 0000000..ce9d3d1 --- /dev/null +++ b/sound/soc/codecs/tas2781.h @@ -0,0 +1,176 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier + * + * Copyright (C) 2022 - 2023 Texas Instruments Incorporated + * https://www.ti.com + * + * The TAS2781 driver implements a flexible and configurable algo coff setting + * for one, two, even multiple TAS2781 chips. + * + * Author: Shenghao Ding <shenghao-ding(a)ti.com> + * Kevin Lu <kevin-lu(a)ti.com> + */ + +#ifndef __TAS2781_H__ +#define __TAS2781_H__ + +#include "tas2781-dsp.h" + +#define SMARTAMP_MODULE_NAME "tas2781" +#define TASDEVICE_RETRY_COUNT 3 +#define TASDEVICE_ERROR_FAILED -2 + +#define TASDEVICE_RATES (SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_88200) +#define TASDEVICE_MAX_CHANNELS 8 + +#define TASDEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/*PAGE Control Register (available in page0 of each book) */ +#define TASDEVICE_PAGE_SELECT 0x00 +#define TASDEVICE_BOOKCTL_PAGE 0x00 +#define TASDEVICE_BOOKCTL_REG 127 +#define TASDEVICE_BOOK_ID(reg) (reg / (256 * 128)) +#define TASDEVICE_PAGE_ID(reg) ((reg % (256 * 128)) / 128) +#define TASDEVICE_PAGE_REG(reg) ((reg % (256 * 128)) % 128) +#define TASDEVICE_PGRG(reg) ((reg % (256 * 128))) +#define TASDEVICE_REG(book, page, reg) (((book * 256 * 128) + \ + (page * 128)) + reg) + + /*Software Reset */ +#define TAS2871_REG_SWRESET TASDEVICE_REG(0x0, 0X0, 0x02) +#define TAS2871_REG_SWRESET_RESET BIT(0) + + /* Enable Global addresses */ +#define TAS2871_MISC_CFG2 TASDEVICE_REG(0x0, 0X0, 0x07) +#define TAS2871_GLOBAL_ADDR_MASK BIT(1) +#define TAS2871_GLOBAL_ADDR_ENABLE BIT(1) + + /*I2C Checksum */ +#define TASDEVICE_I2CChecksum TASDEVICE_REG(0x0, 0x0, 0x7E) + + /* Volume control */ +#define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1A) +#define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) +#define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) + +#define TASDEVICE_CMD_SING_W 0x1 +#define TASDEVICE_CMD_BURST 0x2 +#define TASDEVICE_CMD_DELAY 0x3 +#define TASDEVICE_CMD_FIELD_W 0x4 + +enum audio_device { + TAS2781 = 0, +}; + +#define SMS_HTONS(a, b) ((((a)&0x00FF)<<8) | ((b)&0x00FF)) +#define SMS_HTONL(a, b, c, d) ((((a)&0x000000FF)<<24) | \ + (((b)&0x000000FF)<<16) | (((c)&0x000000FF)<<8) | \ + ((d)&0x000000FF)) + +struct tasdevice { + unsigned int dev_addr; + unsigned int err_code; + short cur_prog; + short cur_conf; + bool is_loading; + bool is_loaderr; + bool is_calibrated_data_loaded; + unsigned char cur_book; + struct tasdevice_fw *cali_data_fmw; +}; + +/* + * This item is used to store the generic i2c address of + * all the tas2781 devices for I2C broadcast during the multi-device + * writes, useless in mono case. + */ +struct global_addr { + unsigned char cur_book; + unsigned int dev_addr; + int ref_cnt; +}; + +struct tas_control { + struct snd_kcontrol_new *tasdevice_profile_controls; + int nr_controls; +}; + +struct tasdevice_irqinfo { + int irq_gpio; + int irq; + struct delayed_work irq_work; + bool is_irq_enable; +}; + +struct tasdevice_priv { + struct device *dev; + void *client; + struct regmap *regmap; + struct mutex codec_lock; + struct mutex dev_lock; + int rst_gpio; + struct tasdevice tasdevice[max_chn]; + struct tasdevice_fw *fmw; + struct tasdevice_rca rcabin; + struct tasdevice_irqinfo irq_info; + struct tas_control tas_ctrl; + struct global_addr glb_addr; + int cur_prog; + int cur_conf; + unsigned int chip_id; + void (*set_global_mode)(struct tasdevice_priv *tas_dev); + int (*fw_parse_variable_header)(struct tasdevice_priv *tas_dev, + const struct firmware *fmw, int offset); + int (*fw_parse_program_data)(struct tasdevice_priv *tas_dev, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); + int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_dev, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); + int (*tasdevice_load_block)(struct tasdevice_priv *tas_dev, + struct tasdev_blk *pBlock); + int (*fw_parse_calibration_data)(struct tasdevice_priv *tas_dev, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); + void (*irq_work_func)(struct tasdevice_priv *tas_dev); + int fw_state; + unsigned int magic_num; + unsigned char ndev; + unsigned char dev_name[32]; + unsigned char rca_binaryname[64]; + unsigned char coef_binaryname[64]; + unsigned char cal_binaryname[max_chn][64]; + bool runtime_suspend; + struct delayed_work tuning_switch_wrk; + void *codec; + int sysclk; + int pstream; + int cstream; + bool is_glb_calibrated_data_loaded; +}; + +int tasdevice_dev_read(struct tasdevice_priv *tasdevice, + enum channel chn, unsigned int reg, unsigned int *value); +int tasdevice_process_block(void *context, + unsigned char *data, unsigned char dev_idx, int sublocksize); +int tasdevice_dev_write(struct tasdevice_priv *tasdevice, + enum channel chn, unsigned int reg, unsigned int value); + +int tasdevice_dev_bulk_write( + struct tasdevice_priv *tasdevice, enum channel chn, + unsigned int reg, unsigned char *p_data, unsigned int n_length); + +int tasdevice_dev_bulk_read(struct tasdevice_priv *tasdevice, + enum channel chn, unsigned int reg, unsigned char *p_data, + unsigned int n_length); + +int tasdevice_dev_update_bits( + struct tasdevice_priv *tasdevice, enum channel chn, + unsigned int reg, unsigned int mask, unsigned int value); + +#endif /*__TAS2781_H__ */ -- 2.34.1

3 2

[PATCH 5.15 415/567] ASoC: kirkwood: Iterate over array indexes instead of using pointer math
by Greg Kroah-Hartman 07 Mar '23

07 Mar '23

From: Kees Cook <keescook(a)chromium.org> [ Upstream commit b3bcedc0402fcdc5c8624c433562d9d1882749d8 ] Walking the dram->cs array was seen as accesses beyond the first array item by the compiler. Instead, use the array index directly. This allows for run-time bounds checking under CONFIG_UBSAN_BOUNDS as well. Seen with GCC 13 with -fstrict-flex-arrays: ../sound/soc/kirkwood/kirkwood-dma.c: In function 'kirkwood_dma_conf_mbus_windows.constprop': ../sound/soc/kirkwood/kirkwood-dma.c:90:24: warning: array subscript 0 is outside array bounds of 'const struct mbus_dram_window[0]' [-Warray-bounds=] 90 | if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { | ~~^~~~~~ Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> Link: https://lore.kernel.org/r/20230127224128.never.410-kees@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index 700a18561a940..640cebd2983e2 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -86,7 +86,7 @@ kirkwood_dma_conf_mbus_windows(void __iomem *base, int win, /* try to find matching cs for current dma address */ for (i = 0; i < dram->num_cs; i++) { - const struct mbus_dram_window *cs = dram->cs + i; + const struct mbus_dram_window *cs = &dram->cs[i]; if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { writel(cs->base & 0xffff0000, base + KIRKWOOD_AUDIO_WIN_BASE_REG(win)); -- 2.39.2

1 0

[PATCH 6.1 623/885] ASoC: kirkwood: Iterate over array indexes instead of using pointer math
by Greg Kroah-Hartman 07 Mar '23

07 Mar '23

From: Kees Cook <keescook(a)chromium.org> [ Upstream commit b3bcedc0402fcdc5c8624c433562d9d1882749d8 ] Walking the dram->cs array was seen as accesses beyond the first array item by the compiler. Instead, use the array index directly. This allows for run-time bounds checking under CONFIG_UBSAN_BOUNDS as well. Seen with GCC 13 with -fstrict-flex-arrays: ../sound/soc/kirkwood/kirkwood-dma.c: In function 'kirkwood_dma_conf_mbus_windows.constprop': ../sound/soc/kirkwood/kirkwood-dma.c:90:24: warning: array subscript 0 is outside array bounds of 'const struct mbus_dram_window[0]' [-Warray-bounds=] 90 | if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { | ~~^~~~~~ Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> Link: https://lore.kernel.org/r/20230127224128.never.410-kees@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index 700a18561a940..640cebd2983e2 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -86,7 +86,7 @@ kirkwood_dma_conf_mbus_windows(void __iomem *base, int win, /* try to find matching cs for current dma address */ for (i = 0; i < dram->num_cs; i++) { - const struct mbus_dram_window *cs = dram->cs + i; + const struct mbus_dram_window *cs = &dram->cs[i]; if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { writel(cs->base & 0xffff0000, base + KIRKWOOD_AUDIO_WIN_BASE_REG(win)); -- 2.39.2

1 0

[PATCH 6.2 0718/1001] ASoC: kirkwood: Iterate over array indexes instead of using pointer math
by Greg Kroah-Hartman 07 Mar '23

07 Mar '23

From: Kees Cook <keescook(a)chromium.org> [ Upstream commit b3bcedc0402fcdc5c8624c433562d9d1882749d8 ] Walking the dram->cs array was seen as accesses beyond the first array item by the compiler. Instead, use the array index directly. This allows for run-time bounds checking under CONFIG_UBSAN_BOUNDS as well. Seen with GCC 13 with -fstrict-flex-arrays: ../sound/soc/kirkwood/kirkwood-dma.c: In function 'kirkwood_dma_conf_mbus_windows.constprop': ../sound/soc/kirkwood/kirkwood-dma.c:90:24: warning: array subscript 0 is outside array bounds of 'const struct mbus_dram_window[0]' [-Warray-bounds=] 90 | if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { | ~~^~~~~~ Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> Link: https://lore.kernel.org/r/20230127224128.never.410-kees@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index 700a18561a940..640cebd2983e2 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -86,7 +86,7 @@ kirkwood_dma_conf_mbus_windows(void __iomem *base, int win, /* try to find matching cs for current dma address */ for (i = 0; i < dram->num_cs; i++) { - const struct mbus_dram_window *cs = dram->cs + i; + const struct mbus_dram_window *cs = &dram->cs[i]; if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { writel(cs->base & 0xffff0000, base + KIRKWOOD_AUDIO_WIN_BASE_REG(win)); -- 2.39.2

1 0

[PATCH 0/5] ASoC: Intel: avs: Machine board fixes
by Amadeusz Sławiński 07 Mar '23

07 Mar '23

Series of adjustments to machine board files. Use fixed format in boards that were not using one. Fix clock handling. Amadeusz Sławiński (3): ASoC: Intel: avs: max98357a: Explicitly define codec format ASoC: Intel: avs: da7219: Explicitly define codec format ASoC: Intel: avs: rt5682: Explicitly define codec format Cezary Rojewski (2): ASoC: Intel: avs: ssm4567: Remove nau8825 bits ASoC: Intel: avs: nau8825: Adjust clock control sound/soc/intel/avs/boards/da7219.c | 21 +++++++++++++++++ sound/soc/intel/avs/boards/max98357a.c | 22 ++++++++++++++++++ sound/soc/intel/avs/boards/nau8825.c | 14 ++++++------ sound/soc/intel/avs/boards/rt5682.c | 22 ++++++++++++++++++ sound/soc/intel/avs/boards/ssm4567.c | 31 -------------------------- 5 files changed, 72 insertions(+), 38 deletions(-) -- 2.34.1

2 6

[bug report] ALSA: hda: cs35l41: Ensure firmware/tuning pairs are always loaded
by Dan Carpenter 07 Mar '23

07 Mar '23

Hello Stefan Binding, The patch cd40dad2ca91: "ALSA: hda: cs35l41: Ensure firmware/tuning pairs are always loaded" from Feb 13, 2023, leads to the following Smatch static checker warning: sound/pci/hda/cs35l41_hda.c:303 cs35l41_request_firmware_files() warn: passing freed memory '*wmfw_firmware' sound/pci/hda/cs35l41_hda.c 291 release_firmware(*wmfw_firmware); 292 kfree(*wmfw_filename); ^^^^^^^^^^^^^^ 293 294 /* fallback try cirrus/part-dspN-fwtype.wmfw */ 295 ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename, ^^^^^^^^^^^^^ Assume kasprintf() fails at the start of the function. Better to set *wmfw_firmware = NULL as the very first line of the function. 296 CS35L41_FIRMWARE_ROOT, NULL, NULL, -1, "wmfw"); 297 if (!ret) 298 /* fallback try cirrus/part-dspN-fwtype.bin */ 299 ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, 300 CS35L41_FIRMWARE_ROOT, NULL, NULL, -1, "bin"); 301 302 if (ret) { --> 303 release_firmware(*wmfw_firmware); ^^^^^^^^^^^^^^ Use after free. 304 kfree(*wmfw_filename); 305 dev_warn(cs35l41->dev, "Unable to find firmware and tuning\n"); 306 } 307 return ret; 308 } regards, dan carpenter

1 0

[PATCH] ASoC: SOF: sof-audio: add support for setting up topology loopback routes
by Peter Ujfalusi 07 Mar '23

07 Mar '23

From: Chao Song <chao.song(a)linux.intel.com> During route setup, playback stream routes are setup by going through sink path, capture stream routes are set up by going through source path. This is not sufficient to handle loopback cases between pipelines configured with different directions, e.g. a sidetone or an amplifier feedback connected to a speaker protection module. So, add the logic to handle such routes between widgets that aren't in the list of DAPM widgets associated with the same PCM. Link: https://github.com/thesofproject/linux/issues/4042 Suggested-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Chao Song <chao.song(a)linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Chao Song <chao.song(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/sof-audio.c | 61 ++++++++++++++++++++++++++++++++++++++- 1 file changed, 60 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 6de388a8d0b8..4f12e137ff63 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -280,9 +280,11 @@ int sof_route_setup(struct snd_sof_dev *sdev, struct snd_soc_dapm_widget *wsourc static int sof_setup_pipeline_connections(struct snd_sof_dev *sdev, struct snd_soc_dapm_widget_list *list, int dir) { + const struct sof_ipc_tplg_ops *tplg_ops = sof_ipc_get_ops(sdev, tplg); struct snd_soc_dapm_widget *widget; + struct snd_sof_route *sroute; struct snd_soc_dapm_path *p; - int ret; + int ret = 0; int i; /* @@ -325,6 +327,63 @@ static int sof_setup_pipeline_connections(struct snd_sof_dev *sdev, } } + /* + * The above loop handles connections between widgets that belong to the DAPM widget list. + * This is not sufficient to handle loopback cases between pipelines configured with + * different directions, e.g. a sidetone or an amplifier feedback connected to a speaker + * protection module. + */ + list_for_each_entry(sroute, &sdev->route_list, list) { + bool src_widget_in_dapm_list, sink_widget_in_dapm_list; + struct snd_sof_widget *swidget; + + if (sroute->setup) + continue; + + src_widget_in_dapm_list = widget_in_list(list, sroute->src_widget->widget); + sink_widget_in_dapm_list = widget_in_list(list, sroute->sink_widget->widget); + + /* + * if both source and sink are in the DAPM list, the route must already have been + * set up above. And if neither are in the DAPM list, the route shouldn't be + * handled now. + */ + if (src_widget_in_dapm_list == sink_widget_in_dapm_list) + continue; + + /* + * At this point either the source widget or the sink widget is in the DAPM list + * with a route that might need to be set up. Check the use_count of the widget + * that is not in the DAPM list to confirm if it is in use currently before setting + * up the route. + */ + if (src_widget_in_dapm_list) + swidget = sroute->sink_widget; + else + swidget = sroute->src_widget; + + mutex_lock(&swidget->setup_mutex); + if (!swidget->use_count) { + mutex_unlock(&swidget->setup_mutex); + continue; + } + + if (tplg_ops && tplg_ops->route_setup) { + /* + * this route will get freed when either the source widget or the sink + * widget is freed during hw_free + */ + ret = tplg_ops->route_setup(sdev, sroute); + if (!ret) + sroute->setup = true; + } + + mutex_unlock(&swidget->setup_mutex); + + if (ret < 0) + return ret; + } + return 0; } -- 2.39.2

1 0

[PATCH] ASoC: soc-core.c: remove useless dev_dbg()
by Kuninori Morimoto 06 Mar '23

06 Mar '23

soc-core.c is using dev_dbg(), but some of them are useless. It indicates many dev_dbg() at snd_soc_runtime_get_dai_fmt(), but all of them are just noise, almost no meanings. dev_dbg() on soc_probe_link_dais() indicates dai link and its loop order, but it is just noise, no information. dev_dbg() on snd_soc_register_dai() is duplicated. This patch cleanup these. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 13 ------------- 1 file changed, 13 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 71b022f7edfd..2faa0d8d0d8e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1033,7 +1033,6 @@ static void snd_soc_runtime_get_dai_fmt(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_dai *dai, *not_used; - struct device *dev = rtd->dev; u64 pos, possible_fmt; unsigned int mask = 0, dai_fmt = 0; int i, j, priority, pri, until; @@ -1075,8 +1074,6 @@ static void snd_soc_runtime_get_dai_fmt(struct snd_soc_pcm_runtime *rtd) */ until = snd_soc_dai_get_fmt_max_priority(rtd); for (priority = 1; priority <= until; priority++) { - - dev_dbg(dev, "priority = %d\n", priority); for_each_rtd_dais(rtd, j, not_used) { possible_fmt = ULLONG_MAX; @@ -1085,7 +1082,6 @@ static void snd_soc_runtime_get_dai_fmt(struct snd_soc_pcm_runtime *rtd) pri = (j >= i) ? priority : priority - 1; fmt = snd_soc_dai_get_fmt(dai, pri); - dev_dbg(dev, "%s: (pri, fmt) = (%d, %016llX)\n", dai->name, pri, fmt); possible_fmt &= fmt; } if (possible_fmt) @@ -1095,8 +1091,6 @@ static void snd_soc_runtime_get_dai_fmt(struct snd_soc_pcm_runtime *rtd) /* Not Found */ return; found: - dev_dbg(dev, "found auto selected format: %016llX\n", possible_fmt); - /* * convert POSSIBLE_DAIFMT to DAIFMT * @@ -1457,11 +1451,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card) for_each_comp_order(order) { for_each_card_rtds(card, rtd) { - - dev_dbg(card->dev, - "ASoC: probe %s dai link %d late %d\n", - card->name, rtd->num, order); - /* probe all rtd connected DAIs in good order */ ret = snd_soc_pcm_dai_probe(rtd, order); if (ret) @@ -2421,8 +2410,6 @@ struct snd_soc_dai *snd_soc_register_dai(struct snd_soc_component *component, struct device *dev = component->dev; struct snd_soc_dai *dai; - dev_dbg(dev, "ASoC: dynamically register DAI %s\n", dev_name(dev)); - lockdep_assert_held(&client_mutex); dai = devm_kzalloc(dev, sizeof(*dai), GFP_KERNEL); -- 2.25.1

2 1

[PATCH v7 0/4] Add CS35L41 shared boost feature
by Lucas Tanure 06 Mar '23

06 Mar '23

Valve's Steam Deck uses CS35L41 in shared boost mode, where both speakers share the boost circuit. Add this support in the shared lib, but for now, shared boost is not supported in HDA systems as would require BIOS changes. Based on David Rhodes shared boost patches. Also, fix boost config overwriting in IRQ found in the review and do a small refactor of the code. Changes from V6: - Fix TX/RX bits in passive amp Changes from V5: - Improved documentation from Cirrrus - All amps use MDSCYN for boost source - Active amp has TX and RX enabled Changes from V4: - Fix Document subject Changes from V3: - Fix wrong code sent - Fix ISO C90 mixed declarations and code Changes from V2: - Drop External boost without VSPK Documentation - Move Shared boost to use values 2 and 3 - Revert back to reg_sequence but reading the value first and only update the necessary bits - Fix bug found by Intel kernel Test Robot Changes from V1: - Fix Documentation patch subject - New patch for External boost without VSPK Documentation - New patch to fix boost IRQ overwriting issue - New patch to refactor IRQ release error code - reinit_completion on pcm_startup - fix DRE switch overwriting - return IRQ_HANDLED in PLL_LOCK case Lucas Tanure (4): ASoC: cs35l41: Only disable internal boost ASoC: cs35l41: Refactor error release code ALSA: cs35l41: Add shared boost feature ASoC: dt-bindings: cirrus,cs35l41: Document CS35l41 shared boost .../bindings/sound/cirrus,cs35l41.yaml | 10 +- include/sound/cs35l41.h | 13 +- sound/pci/hda/cs35l41_hda.c | 6 +- sound/soc/codecs/cs35l41-lib.c | 73 +++++++++- sound/soc/codecs/cs35l41.c | 125 +++++++++--------- sound/soc/codecs/cs35l41.h | 1 + 6 files changed, 157 insertions(+), 71 deletions(-) -- 2.39.2

4 7

[PATCH v7 00/10] Add the PowerQUICC audio support using the QMC
by Herve Codina 06 Mar '23

06 Mar '23

Hi, This series adds support for audio using the QMC controller available in some Freescale PowerQUICC SoCs. This series contains three parts in order to show the different blocks hierarchy and their usage in this support. The first one is related to TSA (Time Slot Assigner). The TSA handles the data present at the pin level (TDM with up to 64 time slots) and dispatchs them to one or more serial controller (SCC). The second is related to QMC (QUICC Multichannel Controller). The QMC handles the data at the serial controller (SCC) level and splits again the data to creates some virtual channels. The last one is related to the audio component (QMC audio). It is the glue between the QMC controller and the ASoC component. It handles one or more QMC virtual channels and creates one DAI per QMC virtual channels handled. Compared to the previous iteration https://lore.kernel.org/linux-kernel/20230217145645.1768659-1-herve.codina@… this v7 series: - remove '#fsl,serial-cells' (TSA) and '#fsl,chan-cells' (QMC) properties - Fix the QMC timeslots mask generation in case of 64 timeslots Best regards, Herve Codina Changes v6 -> v7 - Patch 1 Remove #fsl,serial-cells Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 2, 3, 7, 8 and 10 Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 5 Remove #fsl,chan-cells Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' - Patch 6 Fix the timeslot assigned mask in case of 64 timeslots - Patch 9 Add 'Reviewed-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' Add 'Tested-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' Changes v5 -> v6 - Patch 1 Fix blank lines and spaces Remove fsl,diagnostic-mode Add some maxItems values Renamed fsl,tsa.h to cpm1-fsl,tsa.h - Patch 2 Remove fsl,diagnostic-mode Renamed fsl,tsa.h to cpm1-fsl,tsa.h Add 'Acked-by: Li Yang <leoyang.li(a)nxp.com>' - Patch 3 Renamed fsl,tsa.h to cpm1-fsl,tsa.h - Patch 5 Renamed fsl,tsa.h to cpm1-fsl,tsa.h Add 'Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>' Changes v4 -> v5 - patch 1 Rename fsl,tsa.yaml to fsl,cpm1-tsa.yaml Rename #serial-cells to #fsl,serial-cells and add a description Fix typos Remove examples present in description Use a pattern property for fsl,[rt]x-ts-routes - patch 2 Remove one left out_8() ppc specific function call Remove the no more needed PPC dependency in case of COMPILE_TEST - patch 4 Add 'Acked-by: Michael Ellerman <mpe(a)ellerman.id.au>' - patch 5 Rename fsl,qmc.yaml to fsl,cpm1-scc-qmc.yaml Rename #chan-cells to #fsl,chan-cells and add a description - patch 6 Add the SOC_FSL dependency in case of COMPILE_TEST (issue raised by the kernel test robot). Fix a typo in commit log Add 'Acked-by: Li Yang <leoyang.li(a)nxp.com>' Changes v3 -> v4 - patches 2, 6 and 9 Update code comment format. - patch 1 Fix some description formats. Add 'additionalProperties: false' in subnode. Move fsl,mode to fsl,diagnostic-mode. Change clocks and clock-names properties. Add '#serial-cells' property related to the newly introduced fsl,tsa-serial phandle. - patch 2 Move fsl,mode to fsl,diagnostic-mode. Replace the fsl,tsa phandle and the fsl,tsa-cell-id property by a fsl,tsa-serial phandle and update the related API. Add missing locks. - patch 5 Fix some description format. Replace the fsl,tsa phandle and the fsl,tsa-cell-id property by a fsl,tsa-serial phandle. Rename fsl,mode to fsl,operational-mode and update its description. - patch 6 Replace the fsl,tsa phandle and the fsl,tsa-cell-id property by a fsl,tsa-serial phandle and use the TSA updated API. Rename fsl,mode to fsl,operational-mode. - patch 8 Add 'Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>' Changes v2 -> v3 - All bindings Rename fsl-tsa.h to fsl,tsa.h Add missing vendor prefix Various fixes (quotes, node names, upper/lower case) - patches 1 and 2 (TSA binding specific) Remove 'reserved' values in the routing tables Remove fsl,grant-mode Add a better description for 'fsl,common-rxtx-pins' Fix clocks/clocks-name handling against fsl,common-rxtx-pins Add information related to the delays unit Removed FSL_CPM_TSA_NBCELL Fix license in binding header file fsl,tsa.h - patches 5 and 6 (QMC binding specific) Remove fsl,cpm-command property Add interrupt property constraint - patches 8 and 9 (QMC audio binding specific) Remove 'items' in compatible property definition Add missing 'dai-common.yaml' reference Fix the qmc_chan phandle definition - patch 2 and 6 Use io{read,write}be{32,16} Change commit subjects and logs - patch 4 Add 'Acked-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' Changes v1 -> v2: - patch 2 and 6 Fix kernel test robot errors - other patches No changes Herve Codina (10): dt-bindings: soc: fsl: cpm_qe: Add TSA controller soc: fsl: cpm1: Add support for TSA MAINTAINERS: add the Freescale TSA controller entry powerpc/8xx: Use a larger CPM1 command check mask dt-bindings: soc: fsl: cpm_qe: Add QMC controller soc: fsl: cpm1: Add support for QMC MAINTAINERS: add the Freescale QMC controller entry dt-bindings: sound: Add support for QMC audio ASoC: fsl: Add support for QMC audio MAINTAINERS: add the Freescale QMC audio entry .../soc/fsl/cpm_qe/fsl,cpm1-scc-qmc.yaml | 162 ++ .../bindings/soc/fsl/cpm_qe/fsl,cpm1-tsa.yaml | 205 +++ .../bindings/sound/fsl,qmc-audio.yaml | 117 ++ MAINTAINERS | 25 + arch/powerpc/platforms/8xx/cpm1.c | 2 +- drivers/soc/fsl/qe/Kconfig | 23 + drivers/soc/fsl/qe/Makefile | 2 + drivers/soc/fsl/qe/qmc.c | 1537 +++++++++++++++++ drivers/soc/fsl/qe/tsa.c | 846 +++++++++ drivers/soc/fsl/qe/tsa.h | 42 + include/dt-bindings/soc/cpm1-fsl,tsa.h | 13 + include/soc/fsl/qe/qmc.h | 71 + sound/soc/fsl/Kconfig | 9 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_qmc_audio.c | 735 ++++++++ 15 files changed, 3790 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/soc/fsl/cpm_qe/fsl,cpm1-scc-qmc.yaml create mode 100644 Documentation/devicetree/bindings/soc/fsl/cpm_qe/fsl,cpm1-tsa.yaml create mode 100644 Documentation/devicetree/bindings/sound/fsl,qmc-audio.yaml create mode 100644 drivers/soc/fsl/qe/qmc.c create mode 100644 drivers/soc/fsl/qe/tsa.c create mode 100644 drivers/soc/fsl/qe/tsa.h create mode 100644 include/dt-bindings/soc/cpm1-fsl,tsa.h create mode 100644 include/soc/fsl/qe/qmc.h create mode 100644 sound/soc/fsl/fsl_qmc_audio.c -- 2.39.2

2 11

soundwire: user space test utilities
by Muni Sekhar 06 Mar '23

06 Mar '23

Hi all, I am looking for any existing user space test utilities to test SOUNDWIRE SUBSYSTEM in Linux kernel. Can someone please point me to this. -- Thanks, Sekhar

2 4

[PATCH v6 00/10] Add the PowerQUICC audio support using the QMC
by Herve Codina 06 Mar '23

06 Mar '23

Hi, This series adds support for audio using the QMC controller available in some Freescale PowerQUICC SoCs. This series contains three parts in order to show the different blocks hierarchy and their usage in this support. The first one is related to TSA (Time Slot Assigner). The TSA handles the data present at the pin level (TDM with up to 64 time slots) and dispatchs them to one or more serial controller (SCC). The second is related to QMC (QUICC Multichannel Controller). The QMC handles the data at the serial controller (SCC) level and splits again the data to creates some virtual channels. The last one is related to the audio component (QMC audio). It is the glue between the QMC controller and the ASoC component. It handles one or more QMC virtual channels and creates one DAI per QMC virtual channels handled. Compared to the previous iteration https://lore.kernel.org/linux-kernel/20230216134226.1692107-1-herve.codina@… this v6 series mainly: - fixes bindings Best regards, Herve Codina Changes v5 -> v6 - Patch 1 Fix blank lines and spaces Remove fsl,diagnostic-mode Add some maxItems values Renamed fsl,tsa.h to cpm1-fsl,tsa.h - Patch 2 Remove fsl,diagnostic-mode Renamed fsl,tsa.h to cpm1-fsl,tsa.h Add 'Acked-by: Li Yang <leoyang.li(a)nxp.com>' - Patch 3 Renamed fsl,tsa.h to cpm1-fsl,tsa.h - Patch 5 Renamed fsl,tsa.h to cpm1-fsl,tsa.h Add 'Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>' Changes v4 -> v5 - patch 1 Rename fsl,tsa.yaml to fsl,cpm1-tsa.yaml Rename #serial-cells to #fsl,serial-cells and add a description Fix typos Remove examples present in description Use a pattern property for fsl,[rt]x-ts-routes - patch 2 Remove one left out_8() ppc specific function call Remove the no more needed PPC dependency in case of COMPILE_TEST - patch 4 Add 'Acked-by: Michael Ellerman <mpe(a)ellerman.id.au>' - patch 5 Rename fsl,qmc.yaml to fsl,cpm1-scc-qmc.yaml Rename #chan-cells to #fsl,chan-cells and add a description - patch 6 Add the SOC_FSL dependency in case of COMPILE_TEST (issue raised by the kernel test robot). Fix a typo in commit log Add 'Acked-by: Li Yang <leoyang.li(a)nxp.com>' Changes v3 -> v4 - patches 2, 6 and 9 Update code comment format. - patch 1 Fix some description formats. Add 'additionalProperties: false' in subnode. Move fsl,mode to fsl,diagnostic-mode. Change clocks and clock-names properties. Add '#serial-cells' property related to the newly introduced fsl,tsa-serial phandle. - patch 2 Move fsl,mode to fsl,diagnostic-mode. Replace the fsl,tsa phandle and the fsl,tsa-cell-id property by a fsl,tsa-serial phandle and update the related API. Add missing locks. - patch 5 Fix some description format. Replace the fsl,tsa phandle and the fsl,tsa-cell-id property by a fsl,tsa-serial phandle. Rename fsl,mode to fsl,operational-mode and update its description. - patch 6 Replace the fsl,tsa phandle and the fsl,tsa-cell-id property by a fsl,tsa-serial phandle and use the TSA updated API. Rename fsl,mode to fsl,operational-mode. - patch 8 Add 'Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>' Changes v2 -> v3 - All bindings Rename fsl-tsa.h to fsl,tsa.h Add missing vendor prefix Various fixes (quotes, node names, upper/lower case) - patches 1 and 2 (TSA binding specific) Remove 'reserved' values in the routing tables Remove fsl,grant-mode Add a better description for 'fsl,common-rxtx-pins' Fix clocks/clocks-name handling against fsl,common-rxtx-pins Add information related to the delays unit Removed FSL_CPM_TSA_NBCELL Fix license in binding header file fsl,tsa.h - patches 5 and 6 (QMC binding specific) Remove fsl,cpm-command property Add interrupt property constraint - patches 8 and 9 (QMC audio binding specific) Remove 'items' in compatible property definition Add missing 'dai-common.yaml' reference Fix the qmc_chan phandle definition - patch 2 and 6 Use io{read,write}be{32,16} Change commit subjects and logs - patch 4 Add 'Acked-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' Changes v1 -> v2: - patch 2 and 6 Fix kernel test robot errors - other patches No changes Herve Codina (10): dt-bindings: soc: fsl: cpm_qe: Add TSA controller soc: fsl: cpm1: Add support for TSA MAINTAINERS: add the Freescale TSA controller entry powerpc/8xx: Use a larger CPM1 command check mask dt-bindings: soc: fsl: cpm_qe: Add QMC controller soc: fsl: cpm1: Add support for QMC MAINTAINERS: add the Freescale QMC controller entry dt-bindings: sound: Add support for QMC audio ASoC: fsl: Add support for QMC audio MAINTAINERS: add the Freescale QMC audio entry .../soc/fsl/cpm_qe/fsl,cpm1-scc-qmc.yaml | 172 ++ .../bindings/soc/fsl/cpm_qe/fsl,cpm1-tsa.yaml | 215 +++ .../bindings/sound/fsl,qmc-audio.yaml | 117 ++ MAINTAINERS | 25 + arch/powerpc/platforms/8xx/cpm1.c | 2 +- drivers/soc/fsl/qe/Kconfig | 23 + drivers/soc/fsl/qe/Makefile | 2 + drivers/soc/fsl/qe/qmc.c | 1533 +++++++++++++++++ drivers/soc/fsl/qe/tsa.c | 846 +++++++++ drivers/soc/fsl/qe/tsa.h | 42 + include/dt-bindings/soc/cpm1-fsl,tsa.h | 13 + include/soc/fsl/qe/qmc.h | 71 + sound/soc/fsl/Kconfig | 9 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_qmc_audio.c | 735 ++++++++ 15 files changed, 3806 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/soc/fsl/cpm_qe/fsl,cpm1-scc-qmc.yaml create mode 100644 Documentation/devicetree/bindings/soc/fsl/cpm_qe/fsl,cpm1-tsa.yaml create mode 100644 Documentation/devicetree/bindings/sound/fsl,qmc-audio.yaml create mode 100644 drivers/soc/fsl/qe/qmc.c create mode 100644 drivers/soc/fsl/qe/tsa.c create mode 100644 drivers/soc/fsl/qe/tsa.h create mode 100644 include/dt-bindings/soc/cpm1-fsl,tsa.h create mode 100644 include/soc/fsl/qe/qmc.h create mode 100644 sound/soc/fsl/fsl_qmc_audio.c -- 2.39.1

3 21

sof-hda-dsp: Add speaker led support
by GitHub pull_request - opened 06 Mar '23

06 Mar '23

alsa-project/alsa-ucm-conf pull request #286 was opened from Kobatw: There's a speaker led on Dell G16 and use sof-hda-dsp to control. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/286 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/286.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2] ASoC: Intel: avs: Use struct_size for struct avs_modcfg_ext size
by Jacob Keller 06 Mar '23

06 Mar '23

The struct avs_modcfg_ext structure has a flexible array member for the pin_fmts array, and the size should be calculated using struct_size to prevent the potential for overflow with the allocation. Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Reviewed-by: "Amadeusz Sławiński" <amadeuszx.slawinski(a)linux.intel.com> Signed-off-by: Jacob Keller <jacob.e.keller(a)intel.com> --- sound/soc/intel/avs/path.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/path.c b/sound/soc/intel/avs/path.c index 05302ab705ae..adbe23a47847 100644 --- a/sound/soc/intel/avs/path.c +++ b/sound/soc/intel/avs/path.c @@ -478,7 +478,7 @@ static int avs_modext_create(struct avs_dev *adev, struct avs_path_module *mod) int ret, i; num_pins = tcfg->generic.num_input_pins + tcfg->generic.num_output_pins; - cfg_size = sizeof(*cfg) + sizeof(*cfg->pin_fmts) * num_pins; + cfg_size = struct_size(cfg, pin_fmts, num_pins); cfg = kzalloc(cfg_size, GFP_KERNEL); if (!cfg) base-commit: ee3f96b164688dae21e2466a57f2e806b64e8a37 -- 2.39.1.405.gd4c25cc71f83

2 1

[PATCH 1/2] ASoC: dt-bindings: qcom,wcd934x: Reference dai-common
by Krzysztof Kozlowski 06 Mar '23

06 Mar '23

Reference common DAI properties to get sound-dai-cells description and allow name-prefix. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml index ea09590bfa30..ae8346b9d332 100644 --- a/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml @@ -151,6 +151,7 @@ required: - reg allOf: + - $ref: dai-common.yaml# - if: required: - slim-ifc-dev -- 2.34.1

3 4

[PATCH 0/8] ASoC: microchip: some cleanups for AT91 sound drivers
by Claudiu Beznea 06 Mar '23

06 Mar '23

Hi, The following patches do some cleanups for Microchip AT91 sound drivers. Along with it I took the chance and updated MAINTAINERS file. Thank you, Claudiu Beznea Claudiu Beznea (8): ASoC: mchp-spdiftx: use FIELD_PREP() where possible ASoC: mchp-spdiftx: use regmap_update_bits() ASoC: mchp-spdiftx: update debug message ASoC: mchp-pdmc: use FIELD_PREP() where possible ASoC: mchp-pdmc: return directly ret ASoC: mchp-pdmc: avoid casting to/from void pointer MAINTAINERS: add myself as maintainer for Microchip AT91 sound drivers MAINTAINERS: update Microchip AT91 sound entries with documentation files MAINTAINERS | 9 +++++-- sound/soc/atmel/mchp-pdmc.c | 19 ++++++-------- sound/soc/atmel/mchp-spdiftx.c | 45 ++++++++++++---------------------- 3 files changed, 29 insertions(+), 44 deletions(-) -- 2.34.1

3 11

[PATCH] ASoC: qcom: common: add kcontrol to jack pins
by Srinivas Kandagatla 06 Mar '23

06 Mar '23

Add Kcontrol to jack pins so that device switch in ucm can be done correctly. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/common.c | 17 +++++++++++++++-- 1 file changed, 15 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c index 96fe80241fb4..c1f24af17506 100644 --- a/sound/soc/qcom/common.c +++ b/sound/soc/qcom/common.c @@ -180,6 +180,18 @@ int qcom_snd_parse_of(struct snd_soc_card *card) } EXPORT_SYMBOL_GPL(qcom_snd_parse_of); +static struct snd_soc_jack_pin qcom_headset_jack_pins[] = { + /* Headset */ + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + int qcom_snd_wcd_jack_setup(struct snd_soc_pcm_runtime *rtd, struct snd_soc_jack *jack, bool *jack_setup) { @@ -189,13 +201,14 @@ int qcom_snd_wcd_jack_setup(struct snd_soc_pcm_runtime *rtd, int rval, i; if (!*jack_setup) { - rval = snd_soc_card_jack_new(card, "Headset Jack", + rval = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_LINEOUT | SND_JACK_MECHANICAL | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_BTN_4 | SND_JACK_BTN_5, - jack); + jack, qcom_headset_jack_pins, + ARRAY_SIZE(qcom_headset_jack_pins)); if (rval < 0) { dev_err(card->dev, "Unable to add Headphone Jack\n"); -- 2.21.0

2 1

[PATCH v2 1/2] ASoC: dt-bindings: wlf,wm8524: Convert to json-schema
by Chancel Liu 06 Mar '23

06 Mar '23

Convert the Wolfson WM8524 24-bit 192KHz Stereo DAC device tree binding documentation to json-schema. Signed-off-by: Chancel Liu <chancel.liu(a)nxp.com> --- .../devicetree/bindings/sound/wlf,wm8524.yaml | 40 +++++++++++++++++++ .../devicetree/bindings/sound/wm8524.txt | 16 -------- 2 files changed, 40 insertions(+), 16 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8524.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wm8524.txt diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8524.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8524.yaml new file mode 100644 index 000000000000..4d951ece394e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8524.yaml @@ -0,0 +1,40 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8524.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson WM8524 24-bit 192KHz Stereo DAC + +maintainers: + - patches(a)opensource.cirrus.com + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: wlf,wm8524 + + "#sound-dai-cells": + const: 0 + + wlf,mute-gpios: + maxItems: 1 + description: + a GPIO spec for the MUTE pin. + +required: + - compatible + - wlf,mute-gpios + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + wm8524: codec { + compatible = "wlf,wm8524"; + wlf,mute-gpios = <&gpio1 8 GPIO_ACTIVE_LOW>; + }; diff --git a/Documentation/devicetree/bindings/sound/wm8524.txt b/Documentation/devicetree/bindings/sound/wm8524.txt deleted file mode 100644 index f6c0c263b135..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8524.txt +++ /dev/null @@ -1,16 +0,0 @@ -WM8524 audio CODEC - -This device does not use I2C or SPI but a simple Hardware Control Interface. - -Required properties: - - - compatible : "wlf,wm8524" - - - wlf,mute-gpios: a GPIO spec for the MUTE pin. - -Example: - -wm8524: codec { - compatible = "wlf,wm8524"; - wlf,mute-gpios = <&gpio1 8 GPIO_ACTIVE_LOW>; -}; -- 2.25.1

3 5

[PATCH] ASoC: Intel: avs: Use struct_size for struct avs_modcfg_ext size
by Jacob Keller 06 Mar '23

06 Mar '23

The struct avs_modcfg_ext structure has a flexible array member for the pin_fmts array, and the size should be calculated using struct_size to prevent the potential for overflow with the allocation. Signed-off-by: Jacob Keller <jacob.e.keller(a)intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Reviewed-by: "Amadeusz Sławiński" <amadeuszx.slawinski(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org --- sound/soc/intel/avs/path.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/path.c b/sound/soc/intel/avs/path.c index 05302ab705ae..adbe23a47847 100644 --- a/sound/soc/intel/avs/path.c +++ b/sound/soc/intel/avs/path.c @@ -478,7 +478,7 @@ static int avs_modext_create(struct avs_dev *adev, struct avs_path_module *mod) int ret, i; num_pins = tcfg->generic.num_input_pins + tcfg->generic.num_output_pins; - cfg_size = sizeof(*cfg) + sizeof(*cfg->pin_fmts) * num_pins; + cfg_size = struct_size(cfg, pin_fmts, num_pins); cfg = kzalloc(cfg_size, GFP_KERNEL); if (!cfg) base-commit: ee3f96b164688dae21e2466a57f2e806b64e8a37 -- 2.39.1.405.gd4c25cc71f83

3 3

[PATCH v1 00/10] Fix sound on ASUS Transformers
by Svyatoslav Ryhel 06 Mar '23

06 Mar '23

- add quirk for headset detection used by some T30 devices (ASUS Transformers, LG Optimus 4X HD and Vu); - add RT5631 and MAX9808x machine drivers - add Fortemedia FM34NE DSP driver used by ASUS Transformers and mandatory for correct sound work - bind everything into working configuration David Heidelberg (1): dt-bindings: sound: nvidia,tegra-audio: add RT5631 CODEC Svyatoslav Ryhel (9): dt-bindings: sound: nvidia,tegra-audio-common: add new property sound: soc: jack: allow multiple interrupt per gpio ASoC: tegra: Support RT5631 by machine driver dt-bindings: sound: nvidia,tegra-audio: add MAX9808x CODEC ASoC: tegra: Support MAX9808x by machine driver ARM: tegra: transformers: update bindings of sound graph dt-bindings: dsp: add Fortemedia FM34 DSP staging: dsp: add support for Fortemedia FM34NE DSP ARM: tegra: transformers: bind FM34NE DSP on supported devices .../bindings/dsp/fortemedia,dsp.yaml | 95 ++ .../sound/nvidia,tegra-audio-common.yaml | 4 + .../sound/nvidia,tegra-audio-max9808x.yaml | 93 ++ .../sound/nvidia,tegra-audio-rt5631.yaml | 88 ++ arch/arm/boot/dts/tegra20-asus-tf101.dts | 22 +- arch/arm/boot/dts/tegra30-asus-tf201.dts | 46 + arch/arm/boot/dts/tegra30-asus-tf300t.dts | 34 +- arch/arm/boot/dts/tegra30-asus-tf300tg.dts | 36 + arch/arm/boot/dts/tegra30-asus-tf700t.dts | 36 + .../dts/tegra30-asus-transformer-common.dtsi | 9 +- .../arm/boot/dts/tegra30-pegatron-chagall.dts | 29 +- drivers/staging/Kconfig | 2 + drivers/staging/Makefile | 1 + drivers/staging/dsp/Kconfig | 7 + drivers/staging/dsp/Makefile | 2 + drivers/staging/dsp/dsp-fm34ne.c | 364 ++++++++ drivers/staging/dsp/dsp-fm34ne.h | 845 ++++++++++++++++++ sound/soc/soc-jack.c | 1 + sound/soc/tegra/Kconfig | 18 + sound/soc/tegra/tegra_asoc_machine.c | 120 ++- 20 files changed, 1837 insertions(+), 15 deletions(-) create mode 100644 Documentation/devicetree/bindings/dsp/fortemedia,dsp.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max9808x.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5631.yaml create mode 100644 drivers/staging/dsp/Kconfig create mode 100644 drivers/staging/dsp/Makefile create mode 100644 drivers/staging/dsp/dsp-fm34ne.c create mode 100644 drivers/staging/dsp/dsp-fm34ne.h -- 2.37.2

8 42

[PATCH][next] ASoC: SOF: ipc4-topology: Replace fake flexible arrays with flexible-array member
by Gustavo A. R. Silva 06 Mar '23

06 Mar '23

Zero-length arrays as fake flexible arrays are deprecated and we are moving towards adopting C99 flexible-array members, instead. Use the DECLARE_FLEX_ARRAY() helper macro to transform zero-length arrays in unions with flexible-array members. Address the following warnings found with GCC-13 and -fstrict-flex-arrays=3 enabled: sound/soc/sof/ipc4-control.c:176:77: warning: array subscript i is outside array bounds of ‘struct sof_ipc4_ctrl_value_chan[0]’ [-Warray-bounds=] sound/soc/sof/ipc4-control.c:78:29: warning: array subscript 0 is outside array bounds of ‘struct sof_ipc4_ctrl_value_chan[0]’ [-Warray-bounds=] sound/soc/sof/ipc4-control.c:80:33: warning: array subscript i is outside array bounds of ‘struct sof_ipc4_ctrl_value_chan[0]’ [-Warray-bounds=] sound/soc/sof/ipc4-control.c:95:53: warning: array subscript i is outside array bounds of ‘struct sof_ipc4_ctrl_value_chan[0]’ [-Warray-bounds=] sound/soc/sof/ipc4-control.c:96:53: warning: array subscript i is outside array bounds of ‘struct sof_ipc4_ctrl_value_chan[0]’ [-Warray-bounds=] sound/soc/sof/ipc4-control.c:93:53: warning: array subscript 0 is outside array bounds of ‘struct sof_ipc4_ctrl_value_chan[0]’ [-Warray-bounds=] sound/soc/sof/ipc4-control.c:140:58: warning: array subscript i is outside array bounds of ‘struct sof_ipc4_ctrl_value_chan[0]’ [-Warray-bounds=] sound/soc/sof/ipc4-control.c:141:29: warning: array subscript i is outside array bounds of ‘struct sof_ipc4_ctrl_value_chan[0]’ [-Warray-bounds=] sound/soc/sof/ipc4-control.c:142:29: warning: array subscript i is outside array bounds of ‘struct sof_ipc4_ctrl_value_chan[0]’ [-Warray-bounds=] sound/soc/sof/ipc4-topology.c:1475:36: warning: array subscript i is outside array bounds of ‘struct sof_ipc4_ctrl_value_chan[0]’ [-Warray-bounds=] sound/soc/sof/ipc4-topology.c:1476:36: warning: array subscript i is outside array bounds of ‘struct sof_ipc4_ctrl_value_chan[0]’ [-Warray-bounds=] This helps with the ongoing efforts to tighten the FORTIFY_SOURCE routines on memcpy() and help us make progress towards globally enabling -fstrict-flex-arrays=3 [1]. Link: https://github.com/KSPP/linux/issues/21 Link: https://github.com/KSPP/linux/issues/193 Link: https://github.com/KSPP/linux/issues/258 Link: https://gcc.gnu.org/pipermail/gcc-patches/2022-October/602902.html [1] Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- sound/soc/sof/ipc4-topology.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h index 72529179ac22..8507171a6e4d 100644 --- a/sound/soc/sof/ipc4-topology.h +++ b/sound/soc/sof/ipc4-topology.h @@ -266,8 +266,8 @@ struct sof_ipc4_control_data { int index; union { - struct sof_ipc4_ctrl_value_chan chanv[0]; - struct sof_abi_hdr data[0]; + DECLARE_FLEX_ARRAY(struct sof_ipc4_ctrl_value_chan, chanv); + DECLARE_FLEX_ARRAY(struct sof_abi_hdr, data); }; }; -- 2.34.1

3 2

[PATCH 0/2] Add support for the TAS5733
by Kamel Bouhara 06 Mar '23

06 Mar '23

Hello, This small series extends the tas571x driver to support the TAS5733 audio power amplifier. Kamel Bouhara (2): ASoC: tas571x: add tas5733 compatible ASoC: tas571x: add support for TAS5733 .../devicetree/bindings/sound/tas571x.txt | 1 + sound/soc/codecs/tas571x.c | 59 +++++++++++++++++++ 2 files changed, 60 insertions(+) -- 2.35.1

3 4

[PATCH] ASoC: codecs: tx-macro: Fix for KASAN: slab-out-of-bounds
by Ravulapati Vishnu Vardhan Rao 06 Mar '23

06 Mar '23

When we run syzkaller we get below Out of Bound. "KASAN: slab-out-of-bounds Read in regcache_flat_read" Below is the backtrace of the issue: dump_backtrace+0x0/0x4c8 show_stack+0x34/0x44 dump_stack_lvl+0xd8/0x118 print_address_description+0x30/0x2d8 kasan_report+0x158/0x198 __asan_report_load4_noabort+0x44/0x50 regcache_flat_read+0x10c/0x110 regcache_read+0xf4/0x180 _regmap_read+0xc4/0x278 _regmap_update_bits+0x130/0x290 regmap_update_bits_base+0xc0/0x15c snd_soc_component_update_bits+0xa8/0x22c snd_soc_component_write_field+0x68/0xd4 tx_macro_digital_mute+0xec/0x140 Actually There is no need to have decimator with 32 bits. By limiting the variable with short type u8 issue is resolved. Signed-off-by: Ravulapati Vishnu Vardhan Rao <quic_visr(a)quicinc.com> --- sound/soc/codecs/lpass-tx-macro.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index bf27bdd5be20..473d3cd39554 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -242,7 +242,7 @@ enum { struct tx_mute_work { struct tx_macro *tx; - u32 decimator; + u8 decimator; struct delayed_work dwork; }; @@ -635,7 +635,7 @@ static int tx_macro_mclk_enable(struct tx_macro *tx, return 0; } -static bool is_amic_enabled(struct snd_soc_component *component, int decimator) +static bool is_amic_enabled(struct snd_soc_component *component, u8 decimator) { u16 adc_mux_reg, adc_reg, adc_n; @@ -849,7 +849,7 @@ static int tx_macro_enable_dec(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); - unsigned int decimator; + u8 decimator; u16 tx_vol_ctl_reg, dec_cfg_reg, hpf_gate_reg, tx_gain_ctl_reg; u8 hpf_cut_off_freq; int hpf_delay = TX_MACRO_DMIC_HPF_DELAY_MS; @@ -1064,7 +1064,8 @@ static int tx_macro_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_component *component = dai->component; - u32 decimator, sample_rate; + u32 sample_rate; + u8 decimator; int tx_fs_rate; struct tx_macro *tx = snd_soc_component_get_drvdata(component); @@ -1128,7 +1129,7 @@ static int tx_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream) { struct snd_soc_component *component = dai->component; struct tx_macro *tx = snd_soc_component_get_drvdata(component); - u16 decimator; + u8 decimator; /* active decimator not set yet */ if (tx->active_decimator[dai->id] == -1) -- 2.17.1

2 1

[PATCH] ASoC: clarify that SND_SOC_IMX_SGTL5000 is the old driver
by Luca Ceresoli 06 Mar '23

06 Mar '23

Both SND_SOC_IMX_SGTL5000 and SND_SOC_FSL_ASOC_CARD implement the fsl,imx-audio-sgtl5000 compatible string, which is confusing. It took a little research to find out that the latter is much newer and it is supposed to be the preferred choice since several years. Add a clarification note to avoid wasting time for future readers. Signed-off-by: Luca Ceresoli <luca.ceresoli(a)bootlin.com> --- Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Viorel Suman <viorel.suman(a)nxp.com> Cc: Shengjiu Wang <shengjiu.wang(a)nxp.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- sound/soc/fsl/Kconfig | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 614eceda6b9e..33b67db8794e 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -294,6 +294,10 @@ config SND_SOC_IMX_SGTL5000 Say Y if you want to add support for SoC audio on an i.MX board with a sgtl5000 codec. + Note that this is an old driver. Consider enabling + SND_SOC_FSL_ASOC_CARD and SND_SOC_SGTL5000 to use the newer + driver. + config SND_SOC_IMX_SPDIF tristate "SoC Audio support for i.MX boards with S/PDIF" select SND_SOC_IMX_PCM_DMA -- 2.34.1

2 2

[PATCH] ASoC: qcom: q6prm: fix incorrect clk_root passed to ADSP
by Krzysztof Kozlowski 06 Mar '23

06 Mar '23

The second to last argument is clk_root (root of the clock), however the code called q6prm_request_lpass_clock() with clk_attr instead (copy-paste error). This effectively was passing value of 1 as root clock which worked on some of the SoCs (e.g. SM8450) but fails on others, depending on the ADSP. For example on SM8550 this "1" as root clock is not accepted and results in errors coming from ADSP. Fixes: 2f20640491ed ("ASoC: qdsp6: qdsp6: q6prm: handle clk disable correctly") Cc: <stable(a)vger.kernel.org> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/qcom/qdsp6/q6prm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6prm.c b/sound/soc/qcom/qdsp6/q6prm.c index 8aa1a213bfb7..c1dc5bae715a 100644 --- a/sound/soc/qcom/qdsp6/q6prm.c +++ b/sound/soc/qcom/qdsp6/q6prm.c @@ -183,9 +183,9 @@ int q6prm_set_lpass_clock(struct device *dev, int clk_id, int clk_attr, int clk_ unsigned int freq) { if (freq) - return q6prm_request_lpass_clock(dev, clk_id, clk_attr, clk_attr, freq); + return q6prm_request_lpass_clock(dev, clk_id, clk_attr, clk_root, freq); - return q6prm_release_lpass_clock(dev, clk_id, clk_attr, clk_attr, freq); + return q6prm_release_lpass_clock(dev, clk_id, clk_attr, clk_root, freq); } EXPORT_SYMBOL_GPL(q6prm_set_lpass_clock); -- 2.34.1

3 2

Re: [PATCH] dt-bindings: Fix SPI and I2C bus node names in examples
by Robin van der Gracht 06 Mar '23

06 Mar '23

> From: Rob Herring <robh(a)kernel.org> > Date: Tue, Feb 28, 2023 at 10:54 PM > Subject: [PATCH] dt-bindings: Fix SPI and I2C bus node names in > examples > To: Krzysztof Kozlowski <krzysztof.kozlowski+dt(a)linaro.org> > Cc: <devicetree(a)vger.kernel.org>, Miguel Ojeda <ojeda(a)kernel.org>, > Benson Leung <bleung(a)chromium.org>, Guenter Roeck > <groeck(a)chromium.org>, Stephen Boyd <sboyd(a)kernel.org>, Andrzej Hajda > <andrzej.hajda(a)intel.com>, Neil Armstrong <neil.armstrong(a)linaro.org>, > Robert Foss <rfoss(a)kernel.org>, Thierry Reding > <thierry.reding(a)gmail.com>, Sam Ravnborg <sam(a)ravnborg.org>, MyungJoo > Ham <myungjoo.ham(a)samsung.com>, Chanwoo Choi <cw00.choi(a)samsung.com>, > Linus Walleij <linus.walleij(a)linaro.org>, Bartosz Golaszewski > <brgl(a)bgdev.pl>, Pavel Machek <pavel(a)ucw.cz>, Lee Jones > <lee(a)kernel.org>, Mauro Carvalho Chehab <mchehab(a)kernel.org>, David S. > Miller <davem(a)davemloft.net>, Eric Dumazet <edumazet(a)google.com>, > Jakub Kicinski <kuba(a)kernel.org>, Paolo Abeni <pabeni(a)redhat.com>, > Wolfgang Grandegger <wg(a)grandegger.com>, Kalle Valo > <kvalo(a)kernel.org>, Sebastian Reichel <sre(a)kernel.org>, Mark Brown > <broonie(a)kernel.org>, Greg Kroah-Hartman <gregkh(a)linuxfoundation.org>, > <linux-clk(a)vger.kernel.org>, <dri-devel(a)lists.freedesktop.org>, > <linux-gpio(a)vger.kernel.org>, <linux-i2c(a)vger.kernel.org>, > <linux-leds(a)vger.kernel.org>, <linux-media(a)vger.kernel.org>, > <netdev(a)vger.kernel.org>, <linux-can(a)vger.kernel.org>, > <linux-wireless(a)vger.kernel.org>, <linux-pm(a)vger.kernel.org>, > <alsa-devel(a)alsa-project.org>, <linux-usb(a)vger.kernel.org> > > > SPI and I2C bus node names are expected to be "spi" or "i2c", > respectively, with nothing else, a unit-address, or a '-N' index. A > pattern of 'spi0' or 'i2c0' or similar has crept in. Fix all these > cases. Mostly scripted with the following commands: > > git grep -l '\si2c[0-9] {' Documentation/devicetree/ | xargs sed -i -e > 's/i2c[0-9] {/i2c {/' > git grep -l '\sspi[0-9] {' Documentation/devicetree/ | xargs sed -i -e > 's/spi[0-9] {/spi {/' > > With this, a few errors in examples were exposed and fixed. > > Signed-off-by: Rob Herring <robh(a)kernel.org> > --- > Cc: Miguel Ojeda <ojeda(a)kernel.org> > Cc: Krzysztof Kozlowski <krzysztof.kozlowski+dt(a)linaro.org> > Cc: Benson Leung <bleung(a)chromium.org> > Cc: Guenter Roeck <groeck(a)chromium.org> > Cc: Stephen Boyd <sboyd(a)kernel.org> > Cc: Andrzej Hajda <andrzej.hajda(a)intel.com> > Cc: Neil Armstrong <neil.armstrong(a)linaro.org> > Cc: Robert Foss <rfoss(a)kernel.org> > Cc: Thierry Reding <thierry.reding(a)gmail.com> > Cc: Sam Ravnborg <sam(a)ravnborg.org> > Cc: MyungJoo Ham <myungjoo.ham(a)samsung.com> > Cc: Chanwoo Choi <cw00.choi(a)samsung.com> > Cc: Linus Walleij <linus.walleij(a)linaro.org> > Cc: Bartosz Golaszewski <brgl(a)bgdev.pl> > Cc: Pavel Machek <pavel(a)ucw.cz> > Cc: Lee Jones <lee(a)kernel.org> > Cc: Mauro Carvalho Chehab <mchehab(a)kernel.org> > Cc: "David S. Miller" <davem(a)davemloft.net> > Cc: Eric Dumazet <edumazet(a)google.com> > Cc: Jakub Kicinski <kuba(a)kernel.org> > Cc: Paolo Abeni <pabeni(a)redhat.com> > Cc: Wolfgang Grandegger <wg(a)grandegger.com> > Cc: Kalle Valo <kvalo(a)kernel.org> > Cc: Sebastian Reichel <sre(a)kernel.org> > Cc: Mark Brown <broonie(a)kernel.org> > Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> > Cc: linux-clk(a)vger.kernel.org > Cc: dri-devel(a)lists.freedesktop.org > Cc: linux-gpio(a)vger.kernel.org > Cc: linux-i2c(a)vger.kernel.org > Cc: linux-leds(a)vger.kernel.org > Cc: linux-media(a)vger.kernel.org > Cc: netdev(a)vger.kernel.org > Cc: linux-can(a)vger.kernel.org > Cc: linux-wireless(a)vger.kernel.org > Cc: linux-pm(a)vger.kernel.org > Cc: alsa-devel(a)alsa-project.org > Cc: linux-usb(a)vger.kernel.org > --- > .../bindings/auxdisplay/holtek,ht16k33.yaml | 2 +- .... > 86 files changed, 110 insertions(+), 103 deletions(-) > > diff --git > a/Documentation/devicetree/bindings/auxdisplay/holtek,ht16k33.yaml > b/Documentation/devicetree/bindings/auxdisplay/holtek,ht16k33.yaml > index fc4873deb76f..286e726cd052 100644 > --- a/Documentation/devicetree/bindings/auxdisplay/holtek,ht16k33.yaml > +++ b/Documentation/devicetree/bindings/auxdisplay/holtek,ht16k33.yaml > @@ -72,7 +72,7 @@ examples: > #include <dt-bindings/interrupt-controller/irq.h> > #include <dt-bindings/input/input.h> > #include <dt-bindings/leds/common.h> > - i2c1 { > + i2c { > #address-cells = <1>; > #size-cells = <0>; Acked-by: Robin van der Gracht <robin(a)protonic.nl>

1 0

[PATCH V5 0/8] Add SoundWire support for AMD platforms
by Vijendar Mukunda 06 Mar '23

06 Mar '23

ACP IP(v6.x) block has two SoundWire manager instance support. This patchset adds support for AMD SoundWire manager driver. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> changes since v4: - fix nit-picks in the code. - update naming convention for control word and response buffer vairables. - drop pm_suspend check in prepare callback. - use return statement instead of returning ret variable. changes since v3: - add usleep_range() in command/response implementation. - add usleep_range() in clock stop sequence. - modify usleep_range() values throughout the code. - remove unncessary debug statement from code. - update comment in probe() call. - fix timeout condition checks in the code. - drop "ret" variable in amd_disable_sdw_manager(). Changes since v2: - Remove useless variable initializations. - Add helper function to interpret peripheral status. - Move runtime pm sequence to probe_work workqueue. - Use string "SoundWire" instead of "soundwire" in code. - Update comments in interrupt handler and probe sequence. - Rename "sdw_lock" as "acp_sdw_lock". - Remove __func__ from dev_dbg statements. Changes since v1: - Drop asoc tree based patches. will send asoc patches as a separate series. - Fixed double space errors. - Use dev instead of pci->dev. - Use SoundWire manager terminology. - Remove amd_sdw_compute_slave_ports() function and use exported sdw_compute_slave_ports() function. - Remove unused variable "num_ports" from amd_manager structure. - Drop startup and shutdown dai callbacks. - Drop reset_page_addr callback. - Use relative address offset to program SoundWire manager registers throughout the code. - Separate wake enable interrupt handling from slave status handling logic. - Use acp_mmio to program ACP common registers. - Use dai_runtime_array implementation in dai_ops. - Refactor port_ops callbacks. - Add comments in port_ops callbacks. - Add retry count logic in irq thread to address faulty case. - Add helper function to interpret command response. - Add generic bandwidth allocation dependency in Kconfig options. - Add comments for AMD SoundWire power modes. - Add missing timeout check in amd_init_sdw_manager callback. - Declare frameshape parameters in probe call. - Handle error case in clock stop sequence. - Add comments in pm_prepare and pm_ops callbacks. Vijendar Mukunda (8): soundwire: export sdw_compute_slave_ports() function soundwire: amd: Add support for AMD Manager driver soundwire: amd: register SoundWire manager dai ops soundwire: amd: enable build for AMD SoundWire manager driver soundwire: amd: add SoundWire manager interrupt handling soundwire: amd: add runtime pm ops for AMD SoundWire manager driver soundwire: amd: handle SoundWire wake enable interrupt soundwire: amd: add pm_prepare callback and pm ops support drivers/soundwire/Kconfig | 10 + drivers/soundwire/Makefile | 4 + drivers/soundwire/amd_manager.c | 1324 +++++++++++++++++ drivers/soundwire/amd_manager.h | 275 ++++ drivers/soundwire/bus.h | 9 + .../soundwire/generic_bandwidth_allocation.c | 12 +- include/linux/soundwire/sdw_amd.h | 109 ++ 7 files changed, 1734 insertions(+), 9 deletions(-) create mode 100644 drivers/soundwire/amd_manager.c create mode 100644 drivers/soundwire/amd_manager.h create mode 100644 include/linux/soundwire/sdw_amd.h -- 2.34.1

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[PATCH AUTOSEL 5.10 30/30] soundwire: cadence: Drain the RX FIFO after an IO timeout
by Sasha Levin 03 Mar '23

03 Mar '23

From: Richard Fitzgerald <rf(a)opensource.cirrus.com> [ Upstream commit 0603a47bd3a8f439d7844b841eee1819353063e0 ] If wait_for_completion_timeout() times-out in _cdns_xfer_msg() it is possible that something could have been written to the RX FIFO. In this case, we should drain the RX FIFO so that anything in it doesn't carry over and mess up the next transfer. Obviously, if we got to this state something went wrong, and we don't really know the state of everything. The cleanup in this situation cannot be bullet-proof but we should attempt to avoid breaking future transaction, if only to reduce the amount of error noise when debugging the failure from a kernel log. Note that this patch only implements the draining for blocking (non-deferred) transfers. The deferred API doesn't have any proper handling of error conditions and would need some re-design before implementing cleanup. That is a task for a separate patch... Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20221202161812.4186897-4-rf@opensource.cirrus.com Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/cadence_master.c | 50 ++++++++++++++++-------------- 1 file changed, 27 insertions(+), 23 deletions(-) diff --git a/drivers/soundwire/cadence_master.c b/drivers/soundwire/cadence_master.c index 292c4460eaaa3..18e7d158fcca4 100644 --- a/drivers/soundwire/cadence_master.c +++ b/drivers/soundwire/cadence_master.c @@ -511,6 +511,29 @@ cdns_fill_msg_resp(struct sdw_cdns *cdns, return SDW_CMD_OK; } +static void cdns_read_response(struct sdw_cdns *cdns) +{ + u32 num_resp, cmd_base; + int i; + + /* RX_FIFO_AVAIL can be 2 entries more than the FIFO size */ + BUILD_BUG_ON(ARRAY_SIZE(cdns->response_buf) < CDNS_MCP_CMD_LEN + 2); + + num_resp = cdns_readl(cdns, CDNS_MCP_FIFOSTAT); + num_resp &= CDNS_MCP_RX_FIFO_AVAIL; + if (num_resp > ARRAY_SIZE(cdns->response_buf)) { + dev_warn(cdns->dev, "RX AVAIL %d too long\n", num_resp); + num_resp = ARRAY_SIZE(cdns->response_buf); + } + + cmd_base = CDNS_MCP_CMD_BASE; + + for (i = 0; i < num_resp; i++) { + cdns->response_buf[i] = cdns_readl(cdns, cmd_base); + cmd_base += CDNS_MCP_CMD_WORD_LEN; + } +} + static enum sdw_command_response _cdns_xfer_msg(struct sdw_cdns *cdns, struct sdw_msg *msg, int cmd, int offset, int count, bool defer) @@ -552,6 +575,10 @@ _cdns_xfer_msg(struct sdw_cdns *cdns, struct sdw_msg *msg, int cmd, dev_err(cdns->dev, "IO transfer timed out, cmd %d device %d addr %x len %d\n", cmd, msg->dev_num, msg->addr, msg->len); msg->len = 0; + + /* Drain anything in the RX_FIFO */ + cdns_read_response(cdns); + return SDW_CMD_TIMEOUT; } @@ -720,29 +747,6 @@ EXPORT_SYMBOL(cdns_reset_page_addr); * IRQ handling */ -static void cdns_read_response(struct sdw_cdns *cdns) -{ - u32 num_resp, cmd_base; - int i; - - /* RX_FIFO_AVAIL can be 2 entries more than the FIFO size */ - BUILD_BUG_ON(ARRAY_SIZE(cdns->response_buf) < CDNS_MCP_CMD_LEN + 2); - - num_resp = cdns_readl(cdns, CDNS_MCP_FIFOSTAT); - num_resp &= CDNS_MCP_RX_FIFO_AVAIL; - if (num_resp > ARRAY_SIZE(cdns->response_buf)) { - dev_warn(cdns->dev, "RX AVAIL %d too long\n", num_resp); - num_resp = ARRAY_SIZE(cdns->response_buf); - } - - cmd_base = CDNS_MCP_CMD_BASE; - - for (i = 0; i < num_resp; i++) { - cdns->response_buf[i] = cdns_readl(cdns, cmd_base); - cmd_base += CDNS_MCP_CMD_WORD_LEN; - } -} - static int cdns_update_slave_status(struct sdw_cdns *cdns, u32 slave0, u32 slave1) { -- 2.39.2

1 0

[PATCH AUTOSEL 5.10 29/30] soundwire: cadence: Remove wasted space in response_buf
by Sasha Levin 03 Mar '23

03 Mar '23

From: Richard Fitzgerald <rf(a)opensource.cirrus.com> [ Upstream commit 827c32d0df4bbe0d1c47d79f6a5eabfe9ac75216 ] The response_buf was declared much larger (128 entries) than the number of responses that could ever be written into it. The Cadence IP is configurable up to a maximum of 32 entries, and the datasheet says that RX_FIFO_AVAIL can be 2 larger than this. So allow up to 34 responses. Also add checking in cdns_read_response() to prevent overflowing reponse_buf if RX_FIFO_AVAIL contains an unexpectedly large number. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20221202161812.4186897-3-rf@opensource.cirrus.com Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/cadence_master.c | 7 +++++++ drivers/soundwire/cadence_master.h | 13 ++++++++++++- 2 files changed, 19 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/cadence_master.c b/drivers/soundwire/cadence_master.c index a3247692ddc07..292c4460eaaa3 100644 --- a/drivers/soundwire/cadence_master.c +++ b/drivers/soundwire/cadence_master.c @@ -725,8 +725,15 @@ static void cdns_read_response(struct sdw_cdns *cdns) u32 num_resp, cmd_base; int i; + /* RX_FIFO_AVAIL can be 2 entries more than the FIFO size */ + BUILD_BUG_ON(ARRAY_SIZE(cdns->response_buf) < CDNS_MCP_CMD_LEN + 2); + num_resp = cdns_readl(cdns, CDNS_MCP_FIFOSTAT); num_resp &= CDNS_MCP_RX_FIFO_AVAIL; + if (num_resp > ARRAY_SIZE(cdns->response_buf)) { + dev_warn(cdns->dev, "RX AVAIL %d too long\n", num_resp); + num_resp = ARRAY_SIZE(cdns->response_buf); + } cmd_base = CDNS_MCP_CMD_BASE; diff --git a/drivers/soundwire/cadence_master.h b/drivers/soundwire/cadence_master.h index 4d1aab5b5ec2d..e7f0108d417ca 100644 --- a/drivers/soundwire/cadence_master.h +++ b/drivers/soundwire/cadence_master.h @@ -8,6 +8,12 @@ #define SDW_CADENCE_GSYNC_KHZ 4 /* 4 kHz */ #define SDW_CADENCE_GSYNC_HZ (SDW_CADENCE_GSYNC_KHZ * 1000) +/* + * The Cadence IP supports up to 32 entries in the FIFO, though implementations + * can configure the IP to have a smaller FIFO. + */ +#define CDNS_MCP_IP_MAX_CMD_LEN 32 + /** * struct sdw_cdns_pdi: PDI (Physical Data Interface) instance * @@ -119,7 +125,12 @@ struct sdw_cdns { struct sdw_bus bus; unsigned int instance; - u32 response_buf[0x80]; + /* + * The datasheet says the RX FIFO AVAIL can be 2 entries more + * than the FIFO capacity, so allow for this. + */ + u32 response_buf[CDNS_MCP_IP_MAX_CMD_LEN + 2]; + struct completion tx_complete; struct sdw_defer *defer; -- 2.39.2

1 0

[PATCH AUTOSEL 5.15 50/50] soundwire: cadence: Drain the RX FIFO after an IO timeout
by Sasha Levin 03 Mar '23

03 Mar '23

From: Richard Fitzgerald <rf(a)opensource.cirrus.com> [ Upstream commit 0603a47bd3a8f439d7844b841eee1819353063e0 ] If wait_for_completion_timeout() times-out in _cdns_xfer_msg() it is possible that something could have been written to the RX FIFO. In this case, we should drain the RX FIFO so that anything in it doesn't carry over and mess up the next transfer. Obviously, if we got to this state something went wrong, and we don't really know the state of everything. The cleanup in this situation cannot be bullet-proof but we should attempt to avoid breaking future transaction, if only to reduce the amount of error noise when debugging the failure from a kernel log. Note that this patch only implements the draining for blocking (non-deferred) transfers. The deferred API doesn't have any proper handling of error conditions and would need some re-design before implementing cleanup. That is a task for a separate patch... Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20221202161812.4186897-4-rf@opensource.cirrus.com Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/cadence_master.c | 50 ++++++++++++++++-------------- 1 file changed, 27 insertions(+), 23 deletions(-) diff --git a/drivers/soundwire/cadence_master.c b/drivers/soundwire/cadence_master.c index b229e1e0d7d97..842e08c56796b 100644 --- a/drivers/soundwire/cadence_master.c +++ b/drivers/soundwire/cadence_master.c @@ -555,6 +555,29 @@ cdns_fill_msg_resp(struct sdw_cdns *cdns, return SDW_CMD_OK; } +static void cdns_read_response(struct sdw_cdns *cdns) +{ + u32 num_resp, cmd_base; + int i; + + /* RX_FIFO_AVAIL can be 2 entries more than the FIFO size */ + BUILD_BUG_ON(ARRAY_SIZE(cdns->response_buf) < CDNS_MCP_CMD_LEN + 2); + + num_resp = cdns_readl(cdns, CDNS_MCP_FIFOSTAT); + num_resp &= CDNS_MCP_RX_FIFO_AVAIL; + if (num_resp > ARRAY_SIZE(cdns->response_buf)) { + dev_warn(cdns->dev, "RX AVAIL %d too long\n", num_resp); + num_resp = ARRAY_SIZE(cdns->response_buf); + } + + cmd_base = CDNS_MCP_CMD_BASE; + + for (i = 0; i < num_resp; i++) { + cdns->response_buf[i] = cdns_readl(cdns, cmd_base); + cmd_base += CDNS_MCP_CMD_WORD_LEN; + } +} + static enum sdw_command_response _cdns_xfer_msg(struct sdw_cdns *cdns, struct sdw_msg *msg, int cmd, int offset, int count, bool defer) @@ -596,6 +619,10 @@ _cdns_xfer_msg(struct sdw_cdns *cdns, struct sdw_msg *msg, int cmd, dev_err(cdns->dev, "IO transfer timed out, cmd %d device %d addr %x len %d\n", cmd, msg->dev_num, msg->addr, msg->len); msg->len = 0; + + /* Drain anything in the RX_FIFO */ + cdns_read_response(cdns); + return SDW_CMD_TIMEOUT; } @@ -764,29 +791,6 @@ EXPORT_SYMBOL(cdns_reset_page_addr); * IRQ handling */ -static void cdns_read_response(struct sdw_cdns *cdns) -{ - u32 num_resp, cmd_base; - int i; - - /* RX_FIFO_AVAIL can be 2 entries more than the FIFO size */ - BUILD_BUG_ON(ARRAY_SIZE(cdns->response_buf) < CDNS_MCP_CMD_LEN + 2); - - num_resp = cdns_readl(cdns, CDNS_MCP_FIFOSTAT); - num_resp &= CDNS_MCP_RX_FIFO_AVAIL; - if (num_resp > ARRAY_SIZE(cdns->response_buf)) { - dev_warn(cdns->dev, "RX AVAIL %d too long\n", num_resp); - num_resp = ARRAY_SIZE(cdns->response_buf); - } - - cmd_base = CDNS_MCP_CMD_BASE; - - for (i = 0; i < num_resp; i++) { - cdns->response_buf[i] = cdns_readl(cdns, cmd_base); - cmd_base += CDNS_MCP_CMD_WORD_LEN; - } -} - static int cdns_update_slave_status(struct sdw_cdns *cdns, u64 slave_intstat) { -- 2.39.2

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[PATCH AUTOSEL 5.15 49/50] soundwire: cadence: Remove wasted space in response_buf
by Sasha Levin 03 Mar '23

03 Mar '23

From: Richard Fitzgerald <rf(a)opensource.cirrus.com> [ Upstream commit 827c32d0df4bbe0d1c47d79f6a5eabfe9ac75216 ] The response_buf was declared much larger (128 entries) than the number of responses that could ever be written into it. The Cadence IP is configurable up to a maximum of 32 entries, and the datasheet says that RX_FIFO_AVAIL can be 2 larger than this. So allow up to 34 responses. Also add checking in cdns_read_response() to prevent overflowing reponse_buf if RX_FIFO_AVAIL contains an unexpectedly large number. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20221202161812.4186897-3-rf@opensource.cirrus.com Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/cadence_master.c | 7 +++++++ drivers/soundwire/cadence_master.h | 13 ++++++++++++- 2 files changed, 19 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/cadence_master.c b/drivers/soundwire/cadence_master.c index 18d2f9b3e2010..b229e1e0d7d97 100644 --- a/drivers/soundwire/cadence_master.c +++ b/drivers/soundwire/cadence_master.c @@ -769,8 +769,15 @@ static void cdns_read_response(struct sdw_cdns *cdns) u32 num_resp, cmd_base; int i; + /* RX_FIFO_AVAIL can be 2 entries more than the FIFO size */ + BUILD_BUG_ON(ARRAY_SIZE(cdns->response_buf) < CDNS_MCP_CMD_LEN + 2); + num_resp = cdns_readl(cdns, CDNS_MCP_FIFOSTAT); num_resp &= CDNS_MCP_RX_FIFO_AVAIL; + if (num_resp > ARRAY_SIZE(cdns->response_buf)) { + dev_warn(cdns->dev, "RX AVAIL %d too long\n", num_resp); + num_resp = ARRAY_SIZE(cdns->response_buf); + } cmd_base = CDNS_MCP_CMD_BASE; diff --git a/drivers/soundwire/cadence_master.h b/drivers/soundwire/cadence_master.h index e587aede63bf0..e437a604429fa 100644 --- a/drivers/soundwire/cadence_master.h +++ b/drivers/soundwire/cadence_master.h @@ -8,6 +8,12 @@ #define SDW_CADENCE_GSYNC_KHZ 4 /* 4 kHz */ #define SDW_CADENCE_GSYNC_HZ (SDW_CADENCE_GSYNC_KHZ * 1000) +/* + * The Cadence IP supports up to 32 entries in the FIFO, though implementations + * can configure the IP to have a smaller FIFO. + */ +#define CDNS_MCP_IP_MAX_CMD_LEN 32 + /** * struct sdw_cdns_pdi: PDI (Physical Data Interface) instance * @@ -119,7 +125,12 @@ struct sdw_cdns { struct sdw_bus bus; unsigned int instance; - u32 response_buf[0x80]; + /* + * The datasheet says the RX FIFO AVAIL can be 2 entries more + * than the FIFO capacity, so allow for this. + */ + u32 response_buf[CDNS_MCP_IP_MAX_CMD_LEN + 2]; + struct completion tx_complete; struct sdw_defer *defer; -- 2.39.2

1 0

[PATCH AUTOSEL 5.15 19/50] soundwire: bus_type: Avoid lockdep assert in sdw_drv_probe()
by Sasha Levin 03 Mar '23

03 Mar '23

From: Richard Fitzgerald <rf(a)opensource.cirrus.com> [ Upstream commit 3dca1f89ae3455963d7b53245ecf298ea9bae857 ] Don't hold sdw_dev_lock while calling the peripheral driver probe() and remove() callbacks. Holding sdw_dev_lock around the probe() and remove() calls causes a theoretical mutex inversion which lockdep will assert on. During probe() the sdw_dev_lock mutex is taken first and then ASoC/ALSA locks are taken by the probe() implementation. During normal operation ASoC can take its locks and then trigger a runtime resume of the component. The SoundWire resume will then take sdw_dev_lock. This is the reverse order compared to probe(). It's not necessary to hold sdw_dev_lock when calling the probe() and remove(), it is only used to prevent the bus core calling the driver callbacks if there isn't a driver or the driver is removing. All calls to the driver callbacks are guarded by the 'probed' flag. So if sdw_dev_lock is held while setting and clearing the 'probed' flag this is sufficient to guarantee the safety of callback functions. Removing the mutex from around the call to probe() means that it is now possible for a bus event (PING response) to be handled in parallel with the probe(). But sdw_bus_probe() already has handling for this by calling the device update_status() after the probe() has completed. Example lockdep assert: [ 46.098514] ====================================================== [ 46.104736] WARNING: possible circular locking dependency detected [ 46.110961] 6.1.0-rc4-jamerson #1 Tainted: G E [ 46.116842] ------------------------------------------------------ [ 46.123063] mpg123/1130 is trying to acquire lock: [ 46.127883] ffff8b445031fb80 (&slave->sdw_dev_lock){+.+.}-{3:3}, at: sdw_update_slave_status+0x26/0x70 [ 46.137225] but task is already holding lock: [ 46.143074] ffffffffc1455310 (&card->pcm_mutex){+.+.}-{3:3}, at: dpcm_fe_dai_open+0x49/0x830 [ 46.151536] which lock already depends on the new lock.[ 46.159732] the existing dependency chain (in reverse order) is: [ 46.167231] -> #4 (&card->pcm_mutex){+.+.}-{3:3}: [ 46.173428] __mutex_lock+0x94/0x920 [ 46.177542] snd_soc_dpcm_runtime_update+0x2e/0x100 [ 46.182958] snd_soc_dapm_put_enum_double+0x1c2/0x200 [ 46.188548] snd_ctl_elem_write+0x10c/0x1d0 [ 46.193268] snd_ctl_ioctl+0x126/0x850 [ 46.197556] __x64_sys_ioctl+0x87/0xc0 [ 46.201845] do_syscall_64+0x38/0x90 [ 46.205959] entry_SYSCALL_64_after_hwframe+0x63/0xcd [ 46.211553] -> #3 (&card->controls_rwsem){++++}-{3:3}: [ 46.218188] down_write+0x2b/0xd0 [ 46.222038] snd_ctl_add_replace+0x39/0xb0 [ 46.226672] snd_soc_add_controls+0x53/0x80 [ 46.231393] soc_probe_component+0x1e4/0x2a0 [ 46.236202] snd_soc_bind_card+0x51a/0xc80 [ 46.240836] devm_snd_soc_register_card+0x43/0x90 [ 46.246079] mc_probe+0x982/0xfe0 [snd_soc_sof_sdw] [ 46.251500] platform_probe+0x3c/0xa0 [ 46.255700] really_probe+0xde/0x390 [ 46.259814] __driver_probe_device+0x78/0x180 [ 46.264710] driver_probe_device+0x1e/0x90 [ 46.269347] __driver_attach+0x9f/0x1f0 [ 46.273721] bus_for_each_dev+0x78/0xc0 [ 46.278098] bus_add_driver+0x1ac/0x200 [ 46.282473] driver_register+0x8f/0xf0 [ 46.286759] do_one_initcall+0x58/0x310 [ 46.291136] do_init_module+0x4c/0x1f0 [ 46.295422] __do_sys_finit_module+0xb4/0x130 [ 46.300321] do_syscall_64+0x38/0x90 [ 46.304434] entry_SYSCALL_64_after_hwframe+0x63/0xcd [ 46.310027] -> #2 (&card->mutex){+.+.}-{3:3}: [ 46.315883] __mutex_lock+0x94/0x920 [ 46.320000] snd_soc_bind_card+0x3e/0xc80 [ 46.324551] devm_snd_soc_register_card+0x43/0x90 [ 46.329798] mc_probe+0x982/0xfe0 [snd_soc_sof_sdw] [ 46.335219] platform_probe+0x3c/0xa0 [ 46.339420] really_probe+0xde/0x390 [ 46.343532] __driver_probe_device+0x78/0x180 [ 46.348430] driver_probe_device+0x1e/0x90 [ 46.353065] __driver_attach+0x9f/0x1f0 [ 46.357437] bus_for_each_dev+0x78/0xc0 [ 46.361812] bus_add_driver+0x1ac/0x200 [ 46.366716] driver_register+0x8f/0xf0 [ 46.371528] do_one_initcall+0x58/0x310 [ 46.376424] do_init_module+0x4c/0x1f0 [ 46.381239] __do_sys_finit_module+0xb4/0x130 [ 46.386665] do_syscall_64+0x38/0x90 [ 46.391299] entry_SYSCALL_64_after_hwframe+0x63/0xcd [ 46.397416] -> #1 (client_mutex){+.+.}-{3:3}: [ 46.404307] __mutex_lock+0x94/0x920 [ 46.408941] snd_soc_add_component+0x24/0x2c0 [ 46.414345] devm_snd_soc_register_component+0x54/0xa0 [ 46.420522] cs35l56_common_probe+0x280/0x370 [snd_soc_cs35l56] [ 46.427487] cs35l56_sdw_probe+0xf4/0x170 [snd_soc_cs35l56_sdw] [ 46.434442] sdw_drv_probe+0x80/0x1a0 [ 46.439136] really_probe+0xde/0x390 [ 46.443738] __driver_probe_device+0x78/0x180 [ 46.449120] driver_probe_device+0x1e/0x90 [ 46.454247] __driver_attach+0x9f/0x1f0 [ 46.459106] bus_for_each_dev+0x78/0xc0 [ 46.463971] bus_add_driver+0x1ac/0x200 [ 46.468825] driver_register+0x8f/0xf0 [ 46.473592] do_one_initcall+0x58/0x310 [ 46.478441] do_init_module+0x4c/0x1f0 [ 46.483202] __do_sys_finit_module+0xb4/0x130 [ 46.488572] do_syscall_64+0x38/0x90 [ 46.493158] entry_SYSCALL_64_after_hwframe+0x63/0xcd [ 46.499229] -> #0 (&slave->sdw_dev_lock){+.+.}-{3:3}: [ 46.506737] __lock_acquire+0x1121/0x1df0 [ 46.511765] lock_acquire+0xd5/0x300 [ 46.516360] __mutex_lock+0x94/0x920 [ 46.520949] sdw_update_slave_status+0x26/0x70 [ 46.526409] sdw_clear_slave_status+0xd8/0xe0 [ 46.531783] intel_resume_runtime+0x139/0x2a0 [ 46.537155] __rpm_callback+0x41/0x120 [ 46.541919] rpm_callback+0x5d/0x70 [ 46.546422] rpm_resume+0x531/0x7e0 [ 46.550920] __pm_runtime_resume+0x4a/0x80 [ 46.556024] snd_soc_pcm_component_pm_runtime_get+0x2f/0xc0 [ 46.562611] __soc_pcm_open+0x62/0x520 [ 46.567375] dpcm_be_dai_startup+0x116/0x210 [ 46.572661] dpcm_fe_dai_open+0xf7/0x830 [ 46.577597] snd_pcm_open_substream+0x54a/0x8b0 [ 46.583145] snd_pcm_open.part.0+0xdc/0x200 [ 46.588341] snd_pcm_playback_open+0x51/0x80 [ 46.593625] chrdev_open+0xc0/0x250 [ 46.598129] do_dentry_open+0x15f/0x430 [ 46.602981] path_openat+0x75e/0xa80 [ 46.607575] do_filp_open+0xb2/0x160 [ 46.612162] do_sys_openat2+0x9a/0x160 [ 46.616922] __x64_sys_openat+0x53/0xa0 [ 46.621767] do_syscall_64+0x38/0x90 [ 46.626352] entry_SYSCALL_64_after_hwframe+0x63/0xcd [ 46.632414] other info that might help us debug this:[ 46.641862] Chain exists of: &slave->sdw_dev_lock --> &card->controls_rwsem --> &card->pcm_mutex[ 46.655145] Possible unsafe locking scenario:[ 46.662048] CPU0 CPU1 [ 46.667080] ---- ---- [ 46.672108] lock(&card->pcm_mutex); [ 46.676267] lock(&card->controls_rwsem); [ 46.683382] lock(&card->pcm_mutex); [ 46.690063] lock(&slave->sdw_dev_lock); [ 46.694574] *** DEADLOCK ***[ 46.701942] 2 locks held by mpg123/1130: [ 46.706356] #0: ffff8b4457b22b90 (&pcm->open_mutex){+.+.}-{3:3}, at: snd_pcm_open.part.0+0xc9/0x200 [ 46.715999] #1: ffffffffc1455310 (&card->pcm_mutex){+.+.}-{3:3}, at: dpcm_fe_dai_open+0x49/0x830 [ 46.725390] stack backtrace: [ 46.730752] CPU: 0 PID: 1130 Comm: mpg123 Tainted: G E 6.1.0-rc4-jamerson #1 [ 46.739703] Hardware name: AAEON UP-WHL01/UP-WHL01, BIOS UPW1AM19 11/10/2020 [ 46.747270] Call Trace: [ 46.750239] <TASK> [ 46.752857] dump_stack_lvl+0x56/0x73 [ 46.757045] check_noncircular+0x102/0x120 [ 46.761664] __lock_acquire+0x1121/0x1df0 [ 46.766197] lock_acquire+0xd5/0x300 [ 46.770292] ? sdw_update_slave_status+0x26/0x70 [ 46.775432] ? lock_is_held_type+0xe2/0x140 [ 46.780143] __mutex_lock+0x94/0x920 [ 46.784241] ? sdw_update_slave_status+0x26/0x70 [ 46.789387] ? find_held_lock+0x2b/0x80 [ 46.793750] ? sdw_update_slave_status+0x26/0x70 [ 46.798894] ? lock_release+0x147/0x2f0 [ 46.803262] ? lockdep_init_map_type+0x47/0x250 [ 46.808315] ? sdw_update_slave_status+0x26/0x70 [ 46.813456] sdw_update_slave_status+0x26/0x70 [ 46.818422] sdw_clear_slave_status+0xd8/0xe0 [ 46.823302] ? pm_generic_runtime_suspend+0x30/0x30 [ 46.828706] intel_resume_runtime+0x139/0x2a0 [ 46.833583] ? _raw_spin_unlock_irq+0x24/0x50 [ 46.838462] ? pm_generic_runtime_suspend+0x30/0x30 [ 46.843866] __rpm_callback+0x41/0x120 [ 46.848142] ? pm_generic_runtime_suspend+0x30/0x30 [ 46.853550] rpm_callback+0x5d/0x70 [ 46.857568] rpm_resume+0x531/0x7e0 [ 46.861578] ? _raw_spin_lock_irqsave+0x62/0x70 [ 46.866634] __pm_runtime_resume+0x4a/0x80 [ 46.871258] snd_soc_pcm_component_pm_runtime_get+0x2f/0xc0 [ 46.877358] __soc_pcm_open+0x62/0x520 [ 46.881634] ? dpcm_add_paths.isra.0+0x35d/0x4c0 [ 46.886784] dpcm_be_dai_startup+0x116/0x210 [ 46.891592] dpcm_fe_dai_open+0xf7/0x830 [ 46.896046] ? debug_mutex_init+0x33/0x50 [ 46.900591] snd_pcm_open_substream+0x54a/0x8b0 [ 46.905658] snd_pcm_open.part.0+0xdc/0x200 [ 46.910376] ? wake_up_q+0x90/0x90 [ 46.914312] snd_pcm_playback_open+0x51/0x80 [ 46.919118] chrdev_open+0xc0/0x250 [ 46.923147] ? cdev_device_add+0x90/0x90 [ 46.927608] do_dentry_open+0x15f/0x430 [ 46.931976] path_openat+0x75e/0xa80 [ 46.936086] do_filp_open+0xb2/0x160 [ 46.940194] ? lock_release+0x147/0x2f0 [ 46.944563] ? _raw_spin_unlock+0x29/0x50 [ 46.949101] do_sys_openat2+0x9a/0x160 [ 46.953377] __x64_sys_openat+0x53/0xa0 [ 46.957733] do_syscall_64+0x38/0x90 [ 46.961829] entry_SYSCALL_64_after_hwframe+0x63/0xcd [ 46.967402] RIP: 0033:0x7fa6397ccd3b [ 46.971506] Code: 25 00 00 41 00 3d 00 00 41 00 74 4b 64 8b 04 25 18 00 00 00 85 c0 75 67 44 89 e2 48 89 ee bf 9c ff ff ff b8 01 01 00 00 0f 05 <48> 3d 00 f0 ff ff 0f 87 91 00 00 00 48 8b 4c 24 28 64 48 33 0c 25 [ 46.991413] RSP: 002b:00007fff838e8990 EFLAGS: 00000246 ORIG_RAX: 0000000000000101 [ 46.999580] RAX: ffffffffffffffda RBX: 0000000000080802 RCX: 00007fa6397ccd3b [ 47.007311] RDX: 0000000000080802 RSI: 00007fff838e8b50 RDI: 00000000ffffff9c [ 47.015047] RBP: 00007fff838e8b50 R08: 0000000000000000 R09: 0000000000000011 [ 47.022787] R10: 0000000000000000 R11: 0000000000000246 R12: 0000000000080802 [ 47.030539] R13: 0000000000000004 R14: 0000000000000000 R15: 00007fff838e8b50 [ 47.038289] </TASK> Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20230123172520.339367-1-rf@opensource.cirrus.com Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/bus_type.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/drivers/soundwire/bus_type.c b/drivers/soundwire/bus_type.c index 04b3529f89293..963498db0fd22 100644 --- a/drivers/soundwire/bus_type.c +++ b/drivers/soundwire/bus_type.c @@ -105,20 +105,19 @@ static int sdw_drv_probe(struct device *dev) if (ret) return ret; - mutex_lock(&slave->sdw_dev_lock); - ret = drv->probe(slave, id); if (ret) { name = drv->name; if (!name) name = drv->driver.name; - mutex_unlock(&slave->sdw_dev_lock); dev_err(dev, "Probe of %s failed: %d\n", name, ret); dev_pm_domain_detach(dev, false); return ret; } + mutex_lock(&slave->sdw_dev_lock); + /* device is probed so let's read the properties now */ if (drv->ops && drv->ops->read_prop) drv->ops->read_prop(slave); @@ -167,14 +166,12 @@ static int sdw_drv_remove(struct device *dev) int ret = 0; mutex_lock(&slave->sdw_dev_lock); - slave->probed = false; + mutex_unlock(&slave->sdw_dev_lock); if (drv->remove) ret = drv->remove(slave); - mutex_unlock(&slave->sdw_dev_lock); - dev_pm_domain_detach(dev, false); return ret; -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 60/60] soundwire: cadence: Drain the RX FIFO after an IO timeout
by Sasha Levin 03 Mar '23

03 Mar '23

From: Richard Fitzgerald <rf(a)opensource.cirrus.com> [ Upstream commit 0603a47bd3a8f439d7844b841eee1819353063e0 ] If wait_for_completion_timeout() times-out in _cdns_xfer_msg() it is possible that something could have been written to the RX FIFO. In this case, we should drain the RX FIFO so that anything in it doesn't carry over and mess up the next transfer. Obviously, if we got to this state something went wrong, and we don't really know the state of everything. The cleanup in this situation cannot be bullet-proof but we should attempt to avoid breaking future transaction, if only to reduce the amount of error noise when debugging the failure from a kernel log. Note that this patch only implements the draining for blocking (non-deferred) transfers. The deferred API doesn't have any proper handling of error conditions and would need some re-design before implementing cleanup. That is a task for a separate patch... Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20221202161812.4186897-4-rf@opensource.cirrus.com Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/cadence_master.c | 50 ++++++++++++++++-------------- 1 file changed, 27 insertions(+), 23 deletions(-) diff --git a/drivers/soundwire/cadence_master.c b/drivers/soundwire/cadence_master.c index 4e92a6b1ad626..cb59dc634658c 100644 --- a/drivers/soundwire/cadence_master.c +++ b/drivers/soundwire/cadence_master.c @@ -554,6 +554,29 @@ cdns_fill_msg_resp(struct sdw_cdns *cdns, return SDW_CMD_OK; } +static void cdns_read_response(struct sdw_cdns *cdns) +{ + u32 num_resp, cmd_base; + int i; + + /* RX_FIFO_AVAIL can be 2 entries more than the FIFO size */ + BUILD_BUG_ON(ARRAY_SIZE(cdns->response_buf) < CDNS_MCP_CMD_LEN + 2); + + num_resp = cdns_readl(cdns, CDNS_MCP_FIFOSTAT); + num_resp &= CDNS_MCP_RX_FIFO_AVAIL; + if (num_resp > ARRAY_SIZE(cdns->response_buf)) { + dev_warn(cdns->dev, "RX AVAIL %d too long\n", num_resp); + num_resp = ARRAY_SIZE(cdns->response_buf); + } + + cmd_base = CDNS_MCP_CMD_BASE; + + for (i = 0; i < num_resp; i++) { + cdns->response_buf[i] = cdns_readl(cdns, cmd_base); + cmd_base += CDNS_MCP_CMD_WORD_LEN; + } +} + static enum sdw_command_response _cdns_xfer_msg(struct sdw_cdns *cdns, struct sdw_msg *msg, int cmd, int offset, int count, bool defer) @@ -595,6 +618,10 @@ _cdns_xfer_msg(struct sdw_cdns *cdns, struct sdw_msg *msg, int cmd, dev_err(cdns->dev, "IO transfer timed out, cmd %d device %d addr %x len %d\n", cmd, msg->dev_num, msg->addr, msg->len); msg->len = 0; + + /* Drain anything in the RX_FIFO */ + cdns_read_response(cdns); + return SDW_CMD_TIMEOUT; } @@ -768,29 +795,6 @@ EXPORT_SYMBOL(cdns_read_ping_status); * IRQ handling */ -static void cdns_read_response(struct sdw_cdns *cdns) -{ - u32 num_resp, cmd_base; - int i; - - /* RX_FIFO_AVAIL can be 2 entries more than the FIFO size */ - BUILD_BUG_ON(ARRAY_SIZE(cdns->response_buf) < CDNS_MCP_CMD_LEN + 2); - - num_resp = cdns_readl(cdns, CDNS_MCP_FIFOSTAT); - num_resp &= CDNS_MCP_RX_FIFO_AVAIL; - if (num_resp > ARRAY_SIZE(cdns->response_buf)) { - dev_warn(cdns->dev, "RX AVAIL %d too long\n", num_resp); - num_resp = ARRAY_SIZE(cdns->response_buf); - } - - cmd_base = CDNS_MCP_CMD_BASE; - - for (i = 0; i < num_resp; i++) { - cdns->response_buf[i] = cdns_readl(cdns, cmd_base); - cmd_base += CDNS_MCP_CMD_WORD_LEN; - } -} - static int cdns_update_slave_status(struct sdw_cdns *cdns, u64 slave_intstat) { -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 59/60] soundwire: cadence: Remove wasted space in response_buf
by Sasha Levin 03 Mar '23

03 Mar '23

From: Richard Fitzgerald <rf(a)opensource.cirrus.com> [ Upstream commit 827c32d0df4bbe0d1c47d79f6a5eabfe9ac75216 ] The response_buf was declared much larger (128 entries) than the number of responses that could ever be written into it. The Cadence IP is configurable up to a maximum of 32 entries, and the datasheet says that RX_FIFO_AVAIL can be 2 larger than this. So allow up to 34 responses. Also add checking in cdns_read_response() to prevent overflowing reponse_buf if RX_FIFO_AVAIL contains an unexpectedly large number. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20221202161812.4186897-3-rf@opensource.cirrus.com Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/cadence_master.c | 7 +++++++ drivers/soundwire/cadence_master.h | 13 ++++++++++++- 2 files changed, 19 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/cadence_master.c b/drivers/soundwire/cadence_master.c index 93929f19d0831..4e92a6b1ad626 100644 --- a/drivers/soundwire/cadence_master.c +++ b/drivers/soundwire/cadence_master.c @@ -773,8 +773,15 @@ static void cdns_read_response(struct sdw_cdns *cdns) u32 num_resp, cmd_base; int i; + /* RX_FIFO_AVAIL can be 2 entries more than the FIFO size */ + BUILD_BUG_ON(ARRAY_SIZE(cdns->response_buf) < CDNS_MCP_CMD_LEN + 2); + num_resp = cdns_readl(cdns, CDNS_MCP_FIFOSTAT); num_resp &= CDNS_MCP_RX_FIFO_AVAIL; + if (num_resp > ARRAY_SIZE(cdns->response_buf)) { + dev_warn(cdns->dev, "RX AVAIL %d too long\n", num_resp); + num_resp = ARRAY_SIZE(cdns->response_buf); + } cmd_base = CDNS_MCP_CMD_BASE; diff --git a/drivers/soundwire/cadence_master.h b/drivers/soundwire/cadence_master.h index ca9e805bab88f..51e6ecc027cbc 100644 --- a/drivers/soundwire/cadence_master.h +++ b/drivers/soundwire/cadence_master.h @@ -8,6 +8,12 @@ #define SDW_CADENCE_GSYNC_KHZ 4 /* 4 kHz */ #define SDW_CADENCE_GSYNC_HZ (SDW_CADENCE_GSYNC_KHZ * 1000) +/* + * The Cadence IP supports up to 32 entries in the FIFO, though implementations + * can configure the IP to have a smaller FIFO. + */ +#define CDNS_MCP_IP_MAX_CMD_LEN 32 + /** * struct sdw_cdns_pdi: PDI (Physical Data Interface) instance * @@ -114,7 +120,12 @@ struct sdw_cdns { struct sdw_bus bus; unsigned int instance; - u32 response_buf[0x80]; + /* + * The datasheet says the RX FIFO AVAIL can be 2 entries more + * than the FIFO capacity, so allow for this. + */ + u32 response_buf[CDNS_MCP_IP_MAX_CMD_LEN + 2]; + struct completion tx_complete; struct sdw_defer *defer; -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 23/60] soundwire: bus_type: Avoid lockdep assert in sdw_drv_probe()
by Sasha Levin 03 Mar '23

03 Mar '23

From: Richard Fitzgerald <rf(a)opensource.cirrus.com> [ Upstream commit 3dca1f89ae3455963d7b53245ecf298ea9bae857 ] Don't hold sdw_dev_lock while calling the peripheral driver probe() and remove() callbacks. Holding sdw_dev_lock around the probe() and remove() calls causes a theoretical mutex inversion which lockdep will assert on. During probe() the sdw_dev_lock mutex is taken first and then ASoC/ALSA locks are taken by the probe() implementation. During normal operation ASoC can take its locks and then trigger a runtime resume of the component. The SoundWire resume will then take sdw_dev_lock. This is the reverse order compared to probe(). It's not necessary to hold sdw_dev_lock when calling the probe() and remove(), it is only used to prevent the bus core calling the driver callbacks if there isn't a driver or the driver is removing. All calls to the driver callbacks are guarded by the 'probed' flag. So if sdw_dev_lock is held while setting and clearing the 'probed' flag this is sufficient to guarantee the safety of callback functions. Removing the mutex from around the call to probe() means that it is now possible for a bus event (PING response) to be handled in parallel with the probe(). But sdw_bus_probe() already has handling for this by calling the device update_status() after the probe() has completed. Example lockdep assert: [ 46.098514] ====================================================== [ 46.104736] WARNING: possible circular locking dependency detected [ 46.110961] 6.1.0-rc4-jamerson #1 Tainted: G E [ 46.116842] ------------------------------------------------------ [ 46.123063] mpg123/1130 is trying to acquire lock: [ 46.127883] ffff8b445031fb80 (&slave->sdw_dev_lock){+.+.}-{3:3}, at: sdw_update_slave_status+0x26/0x70 [ 46.137225] but task is already holding lock: [ 46.143074] ffffffffc1455310 (&card->pcm_mutex){+.+.}-{3:3}, at: dpcm_fe_dai_open+0x49/0x830 [ 46.151536] which lock already depends on the new lock.[ 46.159732] the existing dependency chain (in reverse order) is: [ 46.167231] -> #4 (&card->pcm_mutex){+.+.}-{3:3}: [ 46.173428] __mutex_lock+0x94/0x920 [ 46.177542] snd_soc_dpcm_runtime_update+0x2e/0x100 [ 46.182958] snd_soc_dapm_put_enum_double+0x1c2/0x200 [ 46.188548] snd_ctl_elem_write+0x10c/0x1d0 [ 46.193268] snd_ctl_ioctl+0x126/0x850 [ 46.197556] __x64_sys_ioctl+0x87/0xc0 [ 46.201845] do_syscall_64+0x38/0x90 [ 46.205959] entry_SYSCALL_64_after_hwframe+0x63/0xcd [ 46.211553] -> #3 (&card->controls_rwsem){++++}-{3:3}: [ 46.218188] down_write+0x2b/0xd0 [ 46.222038] snd_ctl_add_replace+0x39/0xb0 [ 46.226672] snd_soc_add_controls+0x53/0x80 [ 46.231393] soc_probe_component+0x1e4/0x2a0 [ 46.236202] snd_soc_bind_card+0x51a/0xc80 [ 46.240836] devm_snd_soc_register_card+0x43/0x90 [ 46.246079] mc_probe+0x982/0xfe0 [snd_soc_sof_sdw] [ 46.251500] platform_probe+0x3c/0xa0 [ 46.255700] really_probe+0xde/0x390 [ 46.259814] __driver_probe_device+0x78/0x180 [ 46.264710] driver_probe_device+0x1e/0x90 [ 46.269347] __driver_attach+0x9f/0x1f0 [ 46.273721] bus_for_each_dev+0x78/0xc0 [ 46.278098] bus_add_driver+0x1ac/0x200 [ 46.282473] driver_register+0x8f/0xf0 [ 46.286759] do_one_initcall+0x58/0x310 [ 46.291136] do_init_module+0x4c/0x1f0 [ 46.295422] __do_sys_finit_module+0xb4/0x130 [ 46.300321] do_syscall_64+0x38/0x90 [ 46.304434] entry_SYSCALL_64_after_hwframe+0x63/0xcd [ 46.310027] -> #2 (&card->mutex){+.+.}-{3:3}: [ 46.315883] __mutex_lock+0x94/0x920 [ 46.320000] snd_soc_bind_card+0x3e/0xc80 [ 46.324551] devm_snd_soc_register_card+0x43/0x90 [ 46.329798] mc_probe+0x982/0xfe0 [snd_soc_sof_sdw] [ 46.335219] platform_probe+0x3c/0xa0 [ 46.339420] really_probe+0xde/0x390 [ 46.343532] __driver_probe_device+0x78/0x180 [ 46.348430] driver_probe_device+0x1e/0x90 [ 46.353065] __driver_attach+0x9f/0x1f0 [ 46.357437] bus_for_each_dev+0x78/0xc0 [ 46.361812] bus_add_driver+0x1ac/0x200 [ 46.366716] driver_register+0x8f/0xf0 [ 46.371528] do_one_initcall+0x58/0x310 [ 46.376424] do_init_module+0x4c/0x1f0 [ 46.381239] __do_sys_finit_module+0xb4/0x130 [ 46.386665] do_syscall_64+0x38/0x90 [ 46.391299] entry_SYSCALL_64_after_hwframe+0x63/0xcd [ 46.397416] -> #1 (client_mutex){+.+.}-{3:3}: [ 46.404307] __mutex_lock+0x94/0x920 [ 46.408941] snd_soc_add_component+0x24/0x2c0 [ 46.414345] devm_snd_soc_register_component+0x54/0xa0 [ 46.420522] cs35l56_common_probe+0x280/0x370 [snd_soc_cs35l56] [ 46.427487] cs35l56_sdw_probe+0xf4/0x170 [snd_soc_cs35l56_sdw] [ 46.434442] sdw_drv_probe+0x80/0x1a0 [ 46.439136] really_probe+0xde/0x390 [ 46.443738] __driver_probe_device+0x78/0x180 [ 46.449120] driver_probe_device+0x1e/0x90 [ 46.454247] __driver_attach+0x9f/0x1f0 [ 46.459106] bus_for_each_dev+0x78/0xc0 [ 46.463971] bus_add_driver+0x1ac/0x200 [ 46.468825] driver_register+0x8f/0xf0 [ 46.473592] do_one_initcall+0x58/0x310 [ 46.478441] do_init_module+0x4c/0x1f0 [ 46.483202] __do_sys_finit_module+0xb4/0x130 [ 46.488572] do_syscall_64+0x38/0x90 [ 46.493158] entry_SYSCALL_64_after_hwframe+0x63/0xcd [ 46.499229] -> #0 (&slave->sdw_dev_lock){+.+.}-{3:3}: [ 46.506737] __lock_acquire+0x1121/0x1df0 [ 46.511765] lock_acquire+0xd5/0x300 [ 46.516360] __mutex_lock+0x94/0x920 [ 46.520949] sdw_update_slave_status+0x26/0x70 [ 46.526409] sdw_clear_slave_status+0xd8/0xe0 [ 46.531783] intel_resume_runtime+0x139/0x2a0 [ 46.537155] __rpm_callback+0x41/0x120 [ 46.541919] rpm_callback+0x5d/0x70 [ 46.546422] rpm_resume+0x531/0x7e0 [ 46.550920] __pm_runtime_resume+0x4a/0x80 [ 46.556024] snd_soc_pcm_component_pm_runtime_get+0x2f/0xc0 [ 46.562611] __soc_pcm_open+0x62/0x520 [ 46.567375] dpcm_be_dai_startup+0x116/0x210 [ 46.572661] dpcm_fe_dai_open+0xf7/0x830 [ 46.577597] snd_pcm_open_substream+0x54a/0x8b0 [ 46.583145] snd_pcm_open.part.0+0xdc/0x200 [ 46.588341] snd_pcm_playback_open+0x51/0x80 [ 46.593625] chrdev_open+0xc0/0x250 [ 46.598129] do_dentry_open+0x15f/0x430 [ 46.602981] path_openat+0x75e/0xa80 [ 46.607575] do_filp_open+0xb2/0x160 [ 46.612162] do_sys_openat2+0x9a/0x160 [ 46.616922] __x64_sys_openat+0x53/0xa0 [ 46.621767] do_syscall_64+0x38/0x90 [ 46.626352] entry_SYSCALL_64_after_hwframe+0x63/0xcd [ 46.632414] other info that might help us debug this:[ 46.641862] Chain exists of: &slave->sdw_dev_lock --> &card->controls_rwsem --> &card->pcm_mutex[ 46.655145] Possible unsafe locking scenario:[ 46.662048] CPU0 CPU1 [ 46.667080] ---- ---- [ 46.672108] lock(&card->pcm_mutex); [ 46.676267] lock(&card->controls_rwsem); [ 46.683382] lock(&card->pcm_mutex); [ 46.690063] lock(&slave->sdw_dev_lock); [ 46.694574] *** DEADLOCK ***[ 46.701942] 2 locks held by mpg123/1130: [ 46.706356] #0: ffff8b4457b22b90 (&pcm->open_mutex){+.+.}-{3:3}, at: snd_pcm_open.part.0+0xc9/0x200 [ 46.715999] #1: ffffffffc1455310 (&card->pcm_mutex){+.+.}-{3:3}, at: dpcm_fe_dai_open+0x49/0x830 [ 46.725390] stack backtrace: [ 46.730752] CPU: 0 PID: 1130 Comm: mpg123 Tainted: G E 6.1.0-rc4-jamerson #1 [ 46.739703] Hardware name: AAEON UP-WHL01/UP-WHL01, BIOS UPW1AM19 11/10/2020 [ 46.747270] Call Trace: [ 46.750239] <TASK> [ 46.752857] dump_stack_lvl+0x56/0x73 [ 46.757045] check_noncircular+0x102/0x120 [ 46.761664] __lock_acquire+0x1121/0x1df0 [ 46.766197] lock_acquire+0xd5/0x300 [ 46.770292] ? sdw_update_slave_status+0x26/0x70 [ 46.775432] ? lock_is_held_type+0xe2/0x140 [ 46.780143] __mutex_lock+0x94/0x920 [ 46.784241] ? sdw_update_slave_status+0x26/0x70 [ 46.789387] ? find_held_lock+0x2b/0x80 [ 46.793750] ? sdw_update_slave_status+0x26/0x70 [ 46.798894] ? lock_release+0x147/0x2f0 [ 46.803262] ? lockdep_init_map_type+0x47/0x250 [ 46.808315] ? sdw_update_slave_status+0x26/0x70 [ 46.813456] sdw_update_slave_status+0x26/0x70 [ 46.818422] sdw_clear_slave_status+0xd8/0xe0 [ 46.823302] ? pm_generic_runtime_suspend+0x30/0x30 [ 46.828706] intel_resume_runtime+0x139/0x2a0 [ 46.833583] ? _raw_spin_unlock_irq+0x24/0x50 [ 46.838462] ? pm_generic_runtime_suspend+0x30/0x30 [ 46.843866] __rpm_callback+0x41/0x120 [ 46.848142] ? pm_generic_runtime_suspend+0x30/0x30 [ 46.853550] rpm_callback+0x5d/0x70 [ 46.857568] rpm_resume+0x531/0x7e0 [ 46.861578] ? _raw_spin_lock_irqsave+0x62/0x70 [ 46.866634] __pm_runtime_resume+0x4a/0x80 [ 46.871258] snd_soc_pcm_component_pm_runtime_get+0x2f/0xc0 [ 46.877358] __soc_pcm_open+0x62/0x520 [ 46.881634] ? dpcm_add_paths.isra.0+0x35d/0x4c0 [ 46.886784] dpcm_be_dai_startup+0x116/0x210 [ 46.891592] dpcm_fe_dai_open+0xf7/0x830 [ 46.896046] ? debug_mutex_init+0x33/0x50 [ 46.900591] snd_pcm_open_substream+0x54a/0x8b0 [ 46.905658] snd_pcm_open.part.0+0xdc/0x200 [ 46.910376] ? wake_up_q+0x90/0x90 [ 46.914312] snd_pcm_playback_open+0x51/0x80 [ 46.919118] chrdev_open+0xc0/0x250 [ 46.923147] ? cdev_device_add+0x90/0x90 [ 46.927608] do_dentry_open+0x15f/0x430 [ 46.931976] path_openat+0x75e/0xa80 [ 46.936086] do_filp_open+0xb2/0x160 [ 46.940194] ? lock_release+0x147/0x2f0 [ 46.944563] ? _raw_spin_unlock+0x29/0x50 [ 46.949101] do_sys_openat2+0x9a/0x160 [ 46.953377] __x64_sys_openat+0x53/0xa0 [ 46.957733] do_syscall_64+0x38/0x90 [ 46.961829] entry_SYSCALL_64_after_hwframe+0x63/0xcd [ 46.967402] RIP: 0033:0x7fa6397ccd3b [ 46.971506] Code: 25 00 00 41 00 3d 00 00 41 00 74 4b 64 8b 04 25 18 00 00 00 85 c0 75 67 44 89 e2 48 89 ee bf 9c ff ff ff b8 01 01 00 00 0f 05 <48> 3d 00 f0 ff ff 0f 87 91 00 00 00 48 8b 4c 24 28 64 48 33 0c 25 [ 46.991413] RSP: 002b:00007fff838e8990 EFLAGS: 00000246 ORIG_RAX: 0000000000000101 [ 46.999580] RAX: ffffffffffffffda RBX: 0000000000080802 RCX: 00007fa6397ccd3b [ 47.007311] RDX: 0000000000080802 RSI: 00007fff838e8b50 RDI: 00000000ffffff9c [ 47.015047] RBP: 00007fff838e8b50 R08: 0000000000000000 R09: 0000000000000011 [ 47.022787] R10: 0000000000000000 R11: 0000000000000246 R12: 0000000000080802 [ 47.030539] R13: 0000000000000004 R14: 0000000000000000 R15: 00007fff838e8b50 [ 47.038289] </TASK> Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20230123172520.339367-1-rf@opensource.cirrus.com Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/bus_type.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/drivers/soundwire/bus_type.c b/drivers/soundwire/bus_type.c index 04b3529f89293..963498db0fd22 100644 --- a/drivers/soundwire/bus_type.c +++ b/drivers/soundwire/bus_type.c @@ -105,20 +105,19 @@ static int sdw_drv_probe(struct device *dev) if (ret) return ret; - mutex_lock(&slave->sdw_dev_lock); - ret = drv->probe(slave, id); if (ret) { name = drv->name; if (!name) name = drv->driver.name; - mutex_unlock(&slave->sdw_dev_lock); dev_err(dev, "Probe of %s failed: %d\n", name, ret); dev_pm_domain_detach(dev, false); return ret; } + mutex_lock(&slave->sdw_dev_lock); + /* device is probed so let's read the properties now */ if (drv->ops && drv->ops->read_prop) drv->ops->read_prop(slave); @@ -167,14 +166,12 @@ static int sdw_drv_remove(struct device *dev) int ret = 0; mutex_lock(&slave->sdw_dev_lock); - slave->probed = false; + mutex_unlock(&slave->sdw_dev_lock); if (drv->remove) ret = drv->remove(slave); - mutex_unlock(&slave->sdw_dev_lock); - dev_pm_domain_detach(dev, false); return ret; -- 2.39.2

1 0

[PATCH AUTOSEL 6.2 64/64] soundwire: cadence: Drain the RX FIFO after an IO timeout
by Sasha Levin 03 Mar '23

03 Mar '23

From: Richard Fitzgerald <rf(a)opensource.cirrus.com> [ Upstream commit 0603a47bd3a8f439d7844b841eee1819353063e0 ] If wait_for_completion_timeout() times-out in _cdns_xfer_msg() it is possible that something could have been written to the RX FIFO. In this case, we should drain the RX FIFO so that anything in it doesn't carry over and mess up the next transfer. Obviously, if we got to this state something went wrong, and we don't really know the state of everything. The cleanup in this situation cannot be bullet-proof but we should attempt to avoid breaking future transaction, if only to reduce the amount of error noise when debugging the failure from a kernel log. Note that this patch only implements the draining for blocking (non-deferred) transfers. The deferred API doesn't have any proper handling of error conditions and would need some re-design before implementing cleanup. That is a task for a separate patch... Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20221202161812.4186897-4-rf@opensource.cirrus.com Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/cadence_master.c | 50 ++++++++++++++++-------------- 1 file changed, 27 insertions(+), 23 deletions(-) diff --git a/drivers/soundwire/cadence_master.c b/drivers/soundwire/cadence_master.c index 37eef80939475..901052dfe420a 100644 --- a/drivers/soundwire/cadence_master.c +++ b/drivers/soundwire/cadence_master.c @@ -554,6 +554,29 @@ cdns_fill_msg_resp(struct sdw_cdns *cdns, return SDW_CMD_OK; } +static void cdns_read_response(struct sdw_cdns *cdns) +{ + u32 num_resp, cmd_base; + int i; + + /* RX_FIFO_AVAIL can be 2 entries more than the FIFO size */ + BUILD_BUG_ON(ARRAY_SIZE(cdns->response_buf) < CDNS_MCP_CMD_LEN + 2); + + num_resp = cdns_readl(cdns, CDNS_MCP_FIFOSTAT); + num_resp &= CDNS_MCP_RX_FIFO_AVAIL; + if (num_resp > ARRAY_SIZE(cdns->response_buf)) { + dev_warn(cdns->dev, "RX AVAIL %d too long\n", num_resp); + num_resp = ARRAY_SIZE(cdns->response_buf); + } + + cmd_base = CDNS_MCP_CMD_BASE; + + for (i = 0; i < num_resp; i++) { + cdns->response_buf[i] = cdns_readl(cdns, cmd_base); + cmd_base += CDNS_MCP_CMD_WORD_LEN; + } +} + static enum sdw_command_response _cdns_xfer_msg(struct sdw_cdns *cdns, struct sdw_msg *msg, int cmd, int offset, int count, bool defer) @@ -595,6 +618,10 @@ _cdns_xfer_msg(struct sdw_cdns *cdns, struct sdw_msg *msg, int cmd, dev_err(cdns->dev, "IO transfer timed out, cmd %d device %d addr %x len %d\n", cmd, msg->dev_num, msg->addr, msg->len); msg->len = 0; + + /* Drain anything in the RX_FIFO */ + cdns_read_response(cdns); + return SDW_CMD_TIMEOUT; } @@ -768,29 +795,6 @@ EXPORT_SYMBOL(cdns_read_ping_status); * IRQ handling */ -static void cdns_read_response(struct sdw_cdns *cdns) -{ - u32 num_resp, cmd_base; - int i; - - /* RX_FIFO_AVAIL can be 2 entries more than the FIFO size */ - BUILD_BUG_ON(ARRAY_SIZE(cdns->response_buf) < CDNS_MCP_CMD_LEN + 2); - - num_resp = cdns_readl(cdns, CDNS_MCP_FIFOSTAT); - num_resp &= CDNS_MCP_RX_FIFO_AVAIL; - if (num_resp > ARRAY_SIZE(cdns->response_buf)) { - dev_warn(cdns->dev, "RX AVAIL %d too long\n", num_resp); - num_resp = ARRAY_SIZE(cdns->response_buf); - } - - cmd_base = CDNS_MCP_CMD_BASE; - - for (i = 0; i < num_resp; i++) { - cdns->response_buf[i] = cdns_readl(cdns, cmd_base); - cmd_base += CDNS_MCP_CMD_WORD_LEN; - } -} - static int cdns_update_slave_status(struct sdw_cdns *cdns, u64 slave_intstat) { -- 2.39.2

1 0

[PATCH AUTOSEL 6.2 63/64] soundwire: cadence: Remove wasted space in response_buf
by Sasha Levin 03 Mar '23

03 Mar '23

From: Richard Fitzgerald <rf(a)opensource.cirrus.com> [ Upstream commit 827c32d0df4bbe0d1c47d79f6a5eabfe9ac75216 ] The response_buf was declared much larger (128 entries) than the number of responses that could ever be written into it. The Cadence IP is configurable up to a maximum of 32 entries, and the datasheet says that RX_FIFO_AVAIL can be 2 larger than this. So allow up to 34 responses. Also add checking in cdns_read_response() to prevent overflowing reponse_buf if RX_FIFO_AVAIL contains an unexpectedly large number. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20221202161812.4186897-3-rf@opensource.cirrus.com Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/cadence_master.c | 7 +++++++ drivers/soundwire/cadence_master.h | 13 ++++++++++++- 2 files changed, 19 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/cadence_master.c b/drivers/soundwire/cadence_master.c index a1de363eba3ff..37eef80939475 100644 --- a/drivers/soundwire/cadence_master.c +++ b/drivers/soundwire/cadence_master.c @@ -773,8 +773,15 @@ static void cdns_read_response(struct sdw_cdns *cdns) u32 num_resp, cmd_base; int i; + /* RX_FIFO_AVAIL can be 2 entries more than the FIFO size */ + BUILD_BUG_ON(ARRAY_SIZE(cdns->response_buf) < CDNS_MCP_CMD_LEN + 2); + num_resp = cdns_readl(cdns, CDNS_MCP_FIFOSTAT); num_resp &= CDNS_MCP_RX_FIFO_AVAIL; + if (num_resp > ARRAY_SIZE(cdns->response_buf)) { + dev_warn(cdns->dev, "RX AVAIL %d too long\n", num_resp); + num_resp = ARRAY_SIZE(cdns->response_buf); + } cmd_base = CDNS_MCP_CMD_BASE; diff --git a/drivers/soundwire/cadence_master.h b/drivers/soundwire/cadence_master.h index 0434d70d4b1f5..e0a64b28c6b9c 100644 --- a/drivers/soundwire/cadence_master.h +++ b/drivers/soundwire/cadence_master.h @@ -8,6 +8,12 @@ #define SDW_CADENCE_GSYNC_KHZ 4 /* 4 kHz */ #define SDW_CADENCE_GSYNC_HZ (SDW_CADENCE_GSYNC_KHZ * 1000) +/* + * The Cadence IP supports up to 32 entries in the FIFO, though implementations + * can configure the IP to have a smaller FIFO. + */ +#define CDNS_MCP_IP_MAX_CMD_LEN 32 + /** * struct sdw_cdns_pdi: PDI (Physical Data Interface) instance * @@ -117,7 +123,12 @@ struct sdw_cdns { struct sdw_bus bus; unsigned int instance; - u32 response_buf[0x80]; + /* + * The datasheet says the RX FIFO AVAIL can be 2 entries more + * than the FIFO capacity, so allow for this. + */ + u32 response_buf[CDNS_MCP_IP_MAX_CMD_LEN + 2]; + struct completion tx_complete; struct sdw_defer *defer; -- 2.39.2

1 0

[PATCH AUTOSEL 6.2 26/64] soundwire: bus_type: Avoid lockdep assert in sdw_drv_probe()
by Sasha Levin 03 Mar '23

03 Mar '23

From: Richard Fitzgerald <rf(a)opensource.cirrus.com> [ Upstream commit 3dca1f89ae3455963d7b53245ecf298ea9bae857 ] Don't hold sdw_dev_lock while calling the peripheral driver probe() and remove() callbacks. Holding sdw_dev_lock around the probe() and remove() calls causes a theoretical mutex inversion which lockdep will assert on. During probe() the sdw_dev_lock mutex is taken first and then ASoC/ALSA locks are taken by the probe() implementation. During normal operation ASoC can take its locks and then trigger a runtime resume of the component. The SoundWire resume will then take sdw_dev_lock. This is the reverse order compared to probe(). It's not necessary to hold sdw_dev_lock when calling the probe() and remove(), it is only used to prevent the bus core calling the driver callbacks if there isn't a driver or the driver is removing. All calls to the driver callbacks are guarded by the 'probed' flag. So if sdw_dev_lock is held while setting and clearing the 'probed' flag this is sufficient to guarantee the safety of callback functions. Removing the mutex from around the call to probe() means that it is now possible for a bus event (PING response) to be handled in parallel with the probe(). But sdw_bus_probe() already has handling for this by calling the device update_status() after the probe() has completed. Example lockdep assert: [ 46.098514] ====================================================== [ 46.104736] WARNING: possible circular locking dependency detected [ 46.110961] 6.1.0-rc4-jamerson #1 Tainted: G E [ 46.116842] ------------------------------------------------------ [ 46.123063] mpg123/1130 is trying to acquire lock: [ 46.127883] ffff8b445031fb80 (&slave->sdw_dev_lock){+.+.}-{3:3}, at: sdw_update_slave_status+0x26/0x70 [ 46.137225] but task is already holding lock: [ 46.143074] ffffffffc1455310 (&card->pcm_mutex){+.+.}-{3:3}, at: dpcm_fe_dai_open+0x49/0x830 [ 46.151536] which lock already depends on the new lock.[ 46.159732] the existing dependency chain (in reverse order) is: [ 46.167231] -> #4 (&card->pcm_mutex){+.+.}-{3:3}: [ 46.173428] __mutex_lock+0x94/0x920 [ 46.177542] snd_soc_dpcm_runtime_update+0x2e/0x100 [ 46.182958] snd_soc_dapm_put_enum_double+0x1c2/0x200 [ 46.188548] snd_ctl_elem_write+0x10c/0x1d0 [ 46.193268] snd_ctl_ioctl+0x126/0x850 [ 46.197556] __x64_sys_ioctl+0x87/0xc0 [ 46.201845] do_syscall_64+0x38/0x90 [ 46.205959] entry_SYSCALL_64_after_hwframe+0x63/0xcd [ 46.211553] -> #3 (&card->controls_rwsem){++++}-{3:3}: [ 46.218188] down_write+0x2b/0xd0 [ 46.222038] snd_ctl_add_replace+0x39/0xb0 [ 46.226672] snd_soc_add_controls+0x53/0x80 [ 46.231393] soc_probe_component+0x1e4/0x2a0 [ 46.236202] snd_soc_bind_card+0x51a/0xc80 [ 46.240836] devm_snd_soc_register_card+0x43/0x90 [ 46.246079] mc_probe+0x982/0xfe0 [snd_soc_sof_sdw] [ 46.251500] platform_probe+0x3c/0xa0 [ 46.255700] really_probe+0xde/0x390 [ 46.259814] __driver_probe_device+0x78/0x180 [ 46.264710] driver_probe_device+0x1e/0x90 [ 46.269347] __driver_attach+0x9f/0x1f0 [ 46.273721] bus_for_each_dev+0x78/0xc0 [ 46.278098] bus_add_driver+0x1ac/0x200 [ 46.282473] driver_register+0x8f/0xf0 [ 46.286759] do_one_initcall+0x58/0x310 [ 46.291136] do_init_module+0x4c/0x1f0 [ 46.295422] __do_sys_finit_module+0xb4/0x130 [ 46.300321] do_syscall_64+0x38/0x90 [ 46.304434] entry_SYSCALL_64_after_hwframe+0x63/0xcd [ 46.310027] -> #2 (&card->mutex){+.+.}-{3:3}: [ 46.315883] __mutex_lock+0x94/0x920 [ 46.320000] snd_soc_bind_card+0x3e/0xc80 [ 46.324551] devm_snd_soc_register_card+0x43/0x90 [ 46.329798] mc_probe+0x982/0xfe0 [snd_soc_sof_sdw] [ 46.335219] platform_probe+0x3c/0xa0 [ 46.339420] really_probe+0xde/0x390 [ 46.343532] __driver_probe_device+0x78/0x180 [ 46.348430] driver_probe_device+0x1e/0x90 [ 46.353065] __driver_attach+0x9f/0x1f0 [ 46.357437] bus_for_each_dev+0x78/0xc0 [ 46.361812] bus_add_driver+0x1ac/0x200 [ 46.366716] driver_register+0x8f/0xf0 [ 46.371528] do_one_initcall+0x58/0x310 [ 46.376424] do_init_module+0x4c/0x1f0 [ 46.381239] __do_sys_finit_module+0xb4/0x130 [ 46.386665] do_syscall_64+0x38/0x90 [ 46.391299] entry_SYSCALL_64_after_hwframe+0x63/0xcd [ 46.397416] -> #1 (client_mutex){+.+.}-{3:3}: [ 46.404307] __mutex_lock+0x94/0x920 [ 46.408941] snd_soc_add_component+0x24/0x2c0 [ 46.414345] devm_snd_soc_register_component+0x54/0xa0 [ 46.420522] cs35l56_common_probe+0x280/0x370 [snd_soc_cs35l56] [ 46.427487] cs35l56_sdw_probe+0xf4/0x170 [snd_soc_cs35l56_sdw] [ 46.434442] sdw_drv_probe+0x80/0x1a0 [ 46.439136] really_probe+0xde/0x390 [ 46.443738] __driver_probe_device+0x78/0x180 [ 46.449120] driver_probe_device+0x1e/0x90 [ 46.454247] __driver_attach+0x9f/0x1f0 [ 46.459106] bus_for_each_dev+0x78/0xc0 [ 46.463971] bus_add_driver+0x1ac/0x200 [ 46.468825] driver_register+0x8f/0xf0 [ 46.473592] do_one_initcall+0x58/0x310 [ 46.478441] do_init_module+0x4c/0x1f0 [ 46.483202] __do_sys_finit_module+0xb4/0x130 [ 46.488572] do_syscall_64+0x38/0x90 [ 46.493158] entry_SYSCALL_64_after_hwframe+0x63/0xcd [ 46.499229] -> #0 (&slave->sdw_dev_lock){+.+.}-{3:3}: [ 46.506737] __lock_acquire+0x1121/0x1df0 [ 46.511765] lock_acquire+0xd5/0x300 [ 46.516360] __mutex_lock+0x94/0x920 [ 46.520949] sdw_update_slave_status+0x26/0x70 [ 46.526409] sdw_clear_slave_status+0xd8/0xe0 [ 46.531783] intel_resume_runtime+0x139/0x2a0 [ 46.537155] __rpm_callback+0x41/0x120 [ 46.541919] rpm_callback+0x5d/0x70 [ 46.546422] rpm_resume+0x531/0x7e0 [ 46.550920] __pm_runtime_resume+0x4a/0x80 [ 46.556024] snd_soc_pcm_component_pm_runtime_get+0x2f/0xc0 [ 46.562611] __soc_pcm_open+0x62/0x520 [ 46.567375] dpcm_be_dai_startup+0x116/0x210 [ 46.572661] dpcm_fe_dai_open+0xf7/0x830 [ 46.577597] snd_pcm_open_substream+0x54a/0x8b0 [ 46.583145] snd_pcm_open.part.0+0xdc/0x200 [ 46.588341] snd_pcm_playback_open+0x51/0x80 [ 46.593625] chrdev_open+0xc0/0x250 [ 46.598129] do_dentry_open+0x15f/0x430 [ 46.602981] path_openat+0x75e/0xa80 [ 46.607575] do_filp_open+0xb2/0x160 [ 46.612162] do_sys_openat2+0x9a/0x160 [ 46.616922] __x64_sys_openat+0x53/0xa0 [ 46.621767] do_syscall_64+0x38/0x90 [ 46.626352] entry_SYSCALL_64_after_hwframe+0x63/0xcd [ 46.632414] other info that might help us debug this:[ 46.641862] Chain exists of: &slave->sdw_dev_lock --> &card->controls_rwsem --> &card->pcm_mutex[ 46.655145] Possible unsafe locking scenario:[ 46.662048] CPU0 CPU1 [ 46.667080] ---- ---- [ 46.672108] lock(&card->pcm_mutex); [ 46.676267] lock(&card->controls_rwsem); [ 46.683382] lock(&card->pcm_mutex); [ 46.690063] lock(&slave->sdw_dev_lock); [ 46.694574] *** DEADLOCK ***[ 46.701942] 2 locks held by mpg123/1130: [ 46.706356] #0: ffff8b4457b22b90 (&pcm->open_mutex){+.+.}-{3:3}, at: snd_pcm_open.part.0+0xc9/0x200 [ 46.715999] #1: ffffffffc1455310 (&card->pcm_mutex){+.+.}-{3:3}, at: dpcm_fe_dai_open+0x49/0x830 [ 46.725390] stack backtrace: [ 46.730752] CPU: 0 PID: 1130 Comm: mpg123 Tainted: G E 6.1.0-rc4-jamerson #1 [ 46.739703] Hardware name: AAEON UP-WHL01/UP-WHL01, BIOS UPW1AM19 11/10/2020 [ 46.747270] Call Trace: [ 46.750239] <TASK> [ 46.752857] dump_stack_lvl+0x56/0x73 [ 46.757045] check_noncircular+0x102/0x120 [ 46.761664] __lock_acquire+0x1121/0x1df0 [ 46.766197] lock_acquire+0xd5/0x300 [ 46.770292] ? sdw_update_slave_status+0x26/0x70 [ 46.775432] ? lock_is_held_type+0xe2/0x140 [ 46.780143] __mutex_lock+0x94/0x920 [ 46.784241] ? sdw_update_slave_status+0x26/0x70 [ 46.789387] ? find_held_lock+0x2b/0x80 [ 46.793750] ? sdw_update_slave_status+0x26/0x70 [ 46.798894] ? lock_release+0x147/0x2f0 [ 46.803262] ? lockdep_init_map_type+0x47/0x250 [ 46.808315] ? sdw_update_slave_status+0x26/0x70 [ 46.813456] sdw_update_slave_status+0x26/0x70 [ 46.818422] sdw_clear_slave_status+0xd8/0xe0 [ 46.823302] ? pm_generic_runtime_suspend+0x30/0x30 [ 46.828706] intel_resume_runtime+0x139/0x2a0 [ 46.833583] ? _raw_spin_unlock_irq+0x24/0x50 [ 46.838462] ? pm_generic_runtime_suspend+0x30/0x30 [ 46.843866] __rpm_callback+0x41/0x120 [ 46.848142] ? pm_generic_runtime_suspend+0x30/0x30 [ 46.853550] rpm_callback+0x5d/0x70 [ 46.857568] rpm_resume+0x531/0x7e0 [ 46.861578] ? _raw_spin_lock_irqsave+0x62/0x70 [ 46.866634] __pm_runtime_resume+0x4a/0x80 [ 46.871258] snd_soc_pcm_component_pm_runtime_get+0x2f/0xc0 [ 46.877358] __soc_pcm_open+0x62/0x520 [ 46.881634] ? dpcm_add_paths.isra.0+0x35d/0x4c0 [ 46.886784] dpcm_be_dai_startup+0x116/0x210 [ 46.891592] dpcm_fe_dai_open+0xf7/0x830 [ 46.896046] ? debug_mutex_init+0x33/0x50 [ 46.900591] snd_pcm_open_substream+0x54a/0x8b0 [ 46.905658] snd_pcm_open.part.0+0xdc/0x200 [ 46.910376] ? wake_up_q+0x90/0x90 [ 46.914312] snd_pcm_playback_open+0x51/0x80 [ 46.919118] chrdev_open+0xc0/0x250 [ 46.923147] ? cdev_device_add+0x90/0x90 [ 46.927608] do_dentry_open+0x15f/0x430 [ 46.931976] path_openat+0x75e/0xa80 [ 46.936086] do_filp_open+0xb2/0x160 [ 46.940194] ? lock_release+0x147/0x2f0 [ 46.944563] ? _raw_spin_unlock+0x29/0x50 [ 46.949101] do_sys_openat2+0x9a/0x160 [ 46.953377] __x64_sys_openat+0x53/0xa0 [ 46.957733] do_syscall_64+0x38/0x90 [ 46.961829] entry_SYSCALL_64_after_hwframe+0x63/0xcd [ 46.967402] RIP: 0033:0x7fa6397ccd3b [ 46.971506] Code: 25 00 00 41 00 3d 00 00 41 00 74 4b 64 8b 04 25 18 00 00 00 85 c0 75 67 44 89 e2 48 89 ee bf 9c ff ff ff b8 01 01 00 00 0f 05 <48> 3d 00 f0 ff ff 0f 87 91 00 00 00 48 8b 4c 24 28 64 48 33 0c 25 [ 46.991413] RSP: 002b:00007fff838e8990 EFLAGS: 00000246 ORIG_RAX: 0000000000000101 [ 46.999580] RAX: ffffffffffffffda RBX: 0000000000080802 RCX: 00007fa6397ccd3b [ 47.007311] RDX: 0000000000080802 RSI: 00007fff838e8b50 RDI: 00000000ffffff9c [ 47.015047] RBP: 00007fff838e8b50 R08: 0000000000000000 R09: 0000000000000011 [ 47.022787] R10: 0000000000000000 R11: 0000000000000246 R12: 0000000000080802 [ 47.030539] R13: 0000000000000004 R14: 0000000000000000 R15: 00007fff838e8b50 [ 47.038289] </TASK> Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20230123172520.339367-1-rf@opensource.cirrus.com Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/bus_type.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/drivers/soundwire/bus_type.c b/drivers/soundwire/bus_type.c index 04b3529f89293..963498db0fd22 100644 --- a/drivers/soundwire/bus_type.c +++ b/drivers/soundwire/bus_type.c @@ -105,20 +105,19 @@ static int sdw_drv_probe(struct device *dev) if (ret) return ret; - mutex_lock(&slave->sdw_dev_lock); - ret = drv->probe(slave, id); if (ret) { name = drv->name; if (!name) name = drv->driver.name; - mutex_unlock(&slave->sdw_dev_lock); dev_err(dev, "Probe of %s failed: %d\n", name, ret); dev_pm_domain_detach(dev, false); return ret; } + mutex_lock(&slave->sdw_dev_lock); + /* device is probed so let's read the properties now */ if (drv->ops && drv->ops->read_prop) drv->ops->read_prop(slave); @@ -167,14 +166,12 @@ static int sdw_drv_remove(struct device *dev) int ret = 0; mutex_lock(&slave->sdw_dev_lock); - slave->probed = false; + mutex_unlock(&slave->sdw_dev_lock); if (drv->remove) ret = drv->remove(slave); - mutex_unlock(&slave->sdw_dev_lock); - dev_pm_domain_detach(dev, false); return ret; -- 2.39.2

1 0

[PATCH] ASoC: codecs: tx-macro: Fix for KASAN: slab-out-of-bounds
by Ravulapati Vishnu Vardhan Rao 03 Mar '23

03 Mar '23

From: visr <quic_visr(a)quicinc.com> When we run syzkaller we get below Out of Bound. "KASAN: slab-out-of-bounds Read in regcache_flat_read" Below is the backtrace of the issue: dump_backtrace+0x0/0x4c8 show_stack+0x34/0x44 dump_stack_lvl+0xd8/0x118 print_address_description+0x30/0x2d8 kasan_report+0x158/0x198 __asan_report_load4_noabort+0x44/0x50 regcache_flat_read+0x10c/0x110 regcache_read+0xf4/0x180 _regmap_read+0xc4/0x278 _regmap_update_bits+0x130/0x290 regmap_update_bits_base+0xc0/0x15c snd_soc_component_update_bits+0xa8/0x22c snd_soc_component_write_field+0x68/0xd4 tx_macro_digital_mute+0xec/0x140 Actually There is no need to have decimator with 32 bits. By limiting the variable with short type u8 issue is resolved. Signed-off-by: RAVULAPATI VISHNUVARDHAN RAO <quic_visr(a)quicinc.com> --- sound/soc/codecs/lpass-tx-macro.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index bf27bdd5be20..73014fa6e4b8 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -242,7 +242,7 @@ enum { struct tx_mute_work { struct tx_macro *tx; - u32 decimator; + u8 decimator; struct delayed_work dwork; }; @@ -635,7 +635,7 @@ static int tx_macro_mclk_enable(struct tx_macro *tx, return 0; } -static bool is_amic_enabled(struct snd_soc_component *component, int decimator) +static bool is_amic_enabled(struct snd_soc_component *component, u8 decimator) { u16 adc_mux_reg, adc_reg, adc_n; @@ -849,7 +849,7 @@ static int tx_macro_enable_dec(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); - unsigned int decimator; + u8 decimator; u16 tx_vol_ctl_reg, dec_cfg_reg, hpf_gate_reg, tx_gain_ctl_reg; u8 hpf_cut_off_freq; int hpf_delay = TX_MACRO_DMIC_HPF_DELAY_MS; @@ -1064,9 +1064,10 @@ static int tx_macro_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_component *component = dai->component; - u32 decimator, sample_rate; + u32 sample_rate; int tx_fs_rate; struct tx_macro *tx = snd_soc_component_get_drvdata(component); + u8 decimator = 0; sample_rate = params_rate(params); switch (sample_rate) { @@ -1128,7 +1129,7 @@ static int tx_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream) { struct snd_soc_component *component = dai->component; struct tx_macro *tx = snd_soc_component_get_drvdata(component); - u16 decimator; + u8 decimator; /* active decimator not set yet */ if (tx->active_decimator[dai->id] == -1) -- 2.17.1

4 3

[GIT PULL] ASoC fixes for v6.3
by Mark Brown 03 Mar '23

03 Mar '23

The following changes since commit 76f5aaabce492aa6991c28c96bb78b00b05d06c5: ASoC: soc-ac97: Return correct error codes (2023-02-15 16:09:07 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.3 for you to fetch changes up to b56ec2992a2e43bc3e60d6db86849d31640e791f: ASoC: mediatek: mt8195: add missing initialization (2023-03-01 14:49:17 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.3 Almost all of this is driver specific fixes and new IDs that have come in during the merge window. A good chunk of them are simple ones from me which came about due to a bunch of Mediatek Chromebooks being enabled in KernelCI, there's more where that came from. We do have one small feature added to the PCM core by Claudiu Beznea in order to allow the sequencing required to resolve a noise issue with the Microchip PDMC driver. ---------------------------------------------------------------- Arnd Bergmann (1): ASoC: zl38060 add gpiolib dependency Claudiu Beznea (3): ASoC: soc-pcm: add option to start DMA after DAI ASoC: dt-bindings: sama7g5-pdmc: add microchip,startup-delay-us binding ASoC: mchp-pdmc: fix poc noise at capture startup David Rau (1): ASoC: da7219: Improve the IRQ process to increase the stability Dharageswari.R (1): ASoC: Intel: sof_rt5682: Add quirk for Rex board with mx98360a amplifier Duc Anh Le (1): ASoC: amd: yc: Add DMI entries to support HP OMEN 16-n0xxx (8A43) Geert Uytterhoeven (1): ASoC: rsnd: adg: Fix BRG typos Janne Grunau (1): ASoC: dt-bindings: apple,mca: Add t8112-mca compatible Joseph Hunkeler (1): ASoC: amd: yp: Add OMEN by HP Gaming Laptop 16z-n000 to quirks Kiseok Jo (1): ASoC: SMA1303: Change the value for right output Mark Brown (14): ASoC: mt6358: Fix event generation for wake on voice stage 2 switch ASoC: mt6358: Validate Wake on Voice 2 writes ASoC: mt6358: Remove undefined HPx Mux enumeration values ASoC: mt8183: Remove spammy logging from I2S DAI driver ASoC: mt8183: Fix event generation for I2S DAI operations ASoC: mt8192: Remove spammy log messages ASoC: mt8192: Fix event generation for controls ASoC: mt8192: Report an error if when an invalid sidetone gain is written ASoC: mt8192: Fix range for sidetone positive gain ASoC: sam9g20ek: Disable capture unless building with microphone input ASoC: mt8192: Fixes from initial glance at kselftest ASoC: mt6358: Fixes from an initial glance at a ASoC: mchp-pdmc: fix poc noises when starting ASoC: mt8183: Fixes from an initial glance at a Martin Povišer (3): ASoC: apple: mca: Fix final status read on SERDES reset ASoC: apple: mca: Fix SERDES reset sequence ASoC: apple: mca: Improve handling of unavailable DMA channels Nuno Sá (1): ASoC: adau7118: don't disable regulators on device unbind Trevor Wu (3): ASoC: mediatek: mt8188: correct etdm control return value ASoC: mediatek: mt8188: add missing initialization ASoC: mediatek: mt8195: add missing initialization .../devicetree/bindings/sound/apple,mca.yaml | 1 + .../bindings/sound/microchip,sama7g5-pdmc.yaml | 6 ++ include/sound/soc-component.h | 2 + sound/soc/amd/yc/acp6x-mach.c | 14 +++++ sound/soc/apple/mca.c | 31 ++++++++--- sound/soc/atmel/mchp-pdmc.c | 53 ++++++++++++++++-- sound/soc/atmel/sam9g20_wm8731.c | 3 + sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/adau7118.c | 19 +------ sound/soc/codecs/da7219-aad.c | 41 +++++++++++--- sound/soc/codecs/da7219-aad.h | 4 ++ sound/soc/codecs/mt6358.c | 11 ++-- sound/soc/codecs/sma1303.c | 2 +- sound/soc/intel/boards/sof_rt5682.c | 23 ++++++++ sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 12 ++++ sound/soc/mediatek/mt8183/mt8183-dai-i2s.c | 21 +------ sound/soc/mediatek/mt8188/mt8188-dai-etdm.c | 11 +--- sound/soc/mediatek/mt8192/mt8192-dai-adda.c | 58 +++++++------------- sound/soc/mediatek/mt8195/mt8195-dai-etdm.c | 3 + sound/soc/sh/rcar/adg.c | 64 +++++++++++----------- sound/soc/soc-pcm.c | 27 +++++++-- 21 files changed, 260 insertions(+), 147 deletions(-)

2 1

[PATCH 1/2] ALSA: ice1712: Do not left ice->gpio_mutex locked in aureon_add_controls()
by Dmitry Fomin 03 Mar '23

03 Mar '23

If snd_ctl_add() fails in aureon_add_controls(), it immediately returns and leaves ice->gpio_mutex locked. ice->gpio_mutex locks in snd_ice1712_save_gpio_status and unlocks in snd_ice1712_restore_gpio_status(ice). It seems that the mutex is required only for aureon_cs8415_get(), so snd_ice1712_restore_gpio_status(ice) can be placed just after that. Compile tested only. Found by Linux Verification Center (linuxtesting.org) with SVACE. Fixes: 1da177e4c3f4 ("Linux-2.6.12-rc2") Signed-off-by: Dmitry Fomin <fomindmitriyfoma(a)mail.ru> --- sound/pci/ice1712/aureon.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c index 9a30f6d35d13..40a0e0095030 100644 --- a/sound/pci/ice1712/aureon.c +++ b/sound/pci/ice1712/aureon.c @@ -1892,6 +1892,7 @@ static int aureon_add_controls(struct snd_ice1712 *ice) unsigned char id; snd_ice1712_save_gpio_status(ice); id = aureon_cs8415_get(ice, CS8415_ID); + snd_ice1712_restore_gpio_status(ice); if (id != 0x41) dev_info(ice->card->dev, "No CS8415 chip. Skipping CS8415 controls.\n"); @@ -1909,7 +1910,6 @@ static int aureon_add_controls(struct snd_ice1712 *ice) kctl->id.device = ice->pcm->device; } } - snd_ice1712_restore_gpio_status(ice); } return 0; -- 2.37.2

2 2

[PATCH] ALSA: hda/realtek: Add quirk for HP EliteDesk 800 G6 Tower PC
by Łukasz Stelmach 03 Mar '23

03 Mar '23

HP EliteDesk 800 G6 Tower PC (103c:870c) requires a quirk for enabling headset-mic. Signed-off-by: Łukasz Stelmach <l.stelmach(a)samsung.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) Verified by echo 0x19 0x01a1913c > /sys/class/sound/card0/device/sound/card0/hwC0D0/user_pin_configs echo 1 > /sys/class/sound/card0/device/sound/card0/hwC0D0/reconfig diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6fab7c8fc19a..ac196db573e4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11599,6 +11599,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0698, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x069f, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), + SND_PCI_QUIRK(0x103c, 0x870c, "HP", ALC897_FIXUP_HP_HSMIC_VERB), SND_PCI_QUIRK(0x103c, 0x8719, "HP", ALC897_FIXUP_HP_HSMIC_VERB), SND_PCI_QUIRK(0x103c, 0x873e, "HP", ALC671_FIXUP_HP_HEADSET_MIC2), SND_PCI_QUIRK(0x103c, 0x877e, "HP 288 Pro G6", ALC671_FIXUP_HP_HEADSET_MIC2), -- 2.30.2

3 2

[PATCH] ALSA: hda/realtek: Improve support for Dell Precision 3260
by Jaroslav Kysela 03 Mar '23

03 Mar '23

The headset jack works better with model=alc283-dac-wcaps. Without this option, the headset insertion (separate physical jack) may not be handled correctly (re-insertion is required). It seems that it follows the "Intel Reference Board" defaults. Reported-by: steven_wu2(a)dell.com Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e103bb3693c0..7afbb708b705 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9260,6 +9260,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a62, "Dell Precision 5560", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0a9d, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0a9e, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0ac9, "Dell Precision 3260", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x1028, 0x0b19, "Dell XPS 15 9520", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0b1a, "Dell Precision 5570", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0b37, "Dell Inspiron 16 Plus 7620 2-in-1", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), -- 2.39.0

2 1

[PATCH] ASoC: codecs: tx-macro: Fix for KASAN: slab-out-of-bounds
by Ravulapati Vishnu Vardhan Rao 03 Mar '23

03 Mar '23

From: visr <quic_visr(a)quicinc.com> When we run syzkaller we get below Out of Bound. "KASAN: slab-out-of-bounds Read in regcache_flat_read" Below is the backtrace of the issue: dump_backtrace+0x0/0x4c8 show_stack+0x34/0x44 dump_stack_lvl+0xd8/0x118 print_address_description+0x30/0x2d8 kasan_report+0x158/0x198 __asan_report_load4_noabort+0x44/0x50 regcache_flat_read+0x10c/0x110 regcache_read+0xf4/0x180 _regmap_read+0xc4/0x278 _regmap_update_bits+0x130/0x290 regmap_update_bits_base+0xc0/0x15c snd_soc_component_update_bits+0xa8/0x22c snd_soc_component_write_field+0x68/0xd4 tx_macro_digital_mute+0xec/0x140 Actually There is no need to have decimator with 32 bytes. By limiting the variable with short type u8 issue is resolved. Signed-off-by: visr <quic_visr(a)quicinc.com> --- sound/soc/codecs/lpass-tx-macro.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index bf27bdd5be20..73014fa6e4b8 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -242,7 +242,7 @@ enum { struct tx_mute_work { struct tx_macro *tx; - u32 decimator; + u8 decimator; struct delayed_work dwork; }; @@ -635,7 +635,7 @@ static int tx_macro_mclk_enable(struct tx_macro *tx, return 0; } -static bool is_amic_enabled(struct snd_soc_component *component, int decimator) +static bool is_amic_enabled(struct snd_soc_component *component, u8 decimator) { u16 adc_mux_reg, adc_reg, adc_n; @@ -849,7 +849,7 @@ static int tx_macro_enable_dec(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); - unsigned int decimator; + u8 decimator; u16 tx_vol_ctl_reg, dec_cfg_reg, hpf_gate_reg, tx_gain_ctl_reg; u8 hpf_cut_off_freq; int hpf_delay = TX_MACRO_DMIC_HPF_DELAY_MS; @@ -1064,9 +1064,10 @@ static int tx_macro_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_component *component = dai->component; - u32 decimator, sample_rate; + u32 sample_rate; int tx_fs_rate; struct tx_macro *tx = snd_soc_component_get_drvdata(component); + u8 decimator = 0; sample_rate = params_rate(params); switch (sample_rate) { @@ -1128,7 +1129,7 @@ static int tx_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream) { struct snd_soc_component *component = dai->component; struct tx_macro *tx = snd_soc_component_get_drvdata(component); - u16 decimator; + u8 decimator; /* active decimator not set yet */ if (tx->active_decimator[dai->id] == -1) -- 2.17.1

2 1

[PATCH 0/3] Add "mclk" support for maxim,max9867
by richard.leitner＠linux.dev 03 Mar '23

03 Mar '23

This series adds support for the clocks and clock-names properties in the maxim,max9867 bindings. Furthermore the binding definitions are converted from txt to yaml. The mclk property is needed for one of our boards which uses the the i.MX8MP SAI MCLK as clock for the maxim,max9867. Signed-off-by: Richard Leitner <richard.leitner(a)skidata.com> --- Benjamin Bara (1): ASoC: maxim,max9867: add "mclk" support Richard Leitner (2): ASoC: dt-bindings: maxim,max9867: convert txt bindings to yaml ASoC: dt-bindings: maxim,max9867: add "mclk" property .../devicetree/bindings/sound/max9867.txt | 17 ------ .../devicetree/bindings/sound/maxim,max9867.yaml | 61 ++++++++++++++++++++++ sound/soc/codecs/max9867.c | 14 ++++- 3 files changed, 74 insertions(+), 18 deletions(-) --- base-commit: c9c3395d5e3dcc6daee66c6908354d47bf98cb0c change-id: 20230302-max9867-49081908a2ab Best regards, -- Richard Leitner <richard.leitner(a)skidata.com>

6 14

[regression] Bug 217084 - Roland VS-100 non working (snd_usb_audio) since kernel 5.11
by Linux regression tracking (Thorsten Leemhuis) 03 Mar '23

03 Mar '23

Hi, this is your Linux kernel regression tracker. I noticed a regression report in bugzilla.kernel.org. As many (most?) kernel developer don't keep an eye on it, I decided to forward it by mail. Quoting from https://bugzilla.kernel.org/show_bug.cgi?id=217084 : > Alberto Zin 2023-02-24 18:11:31 UTC > > The issue I encountered is on a Roland VS-100: > > https://www.roland.com/global/products/v-studio_100/ > > it seems that snd_usb_audio module has broken the compatibility with the VS-100 at around kernel 5.11 time. After that kernel the interface is recognized only as a midi interface, on recent kernels there is only dummy audio device and no sound at all. > > I have two alsa-info reports, on the same desktop pc: > > working (my current setup, linux kernel 5.6.0-1055-oem): > http://alsa-project.org/db/?f=a3378ecb0a258b4745827c693f4c0045fdb83847 > > non working (live, avlinux, 6.0.x): > http://alsa-project.org/db/?f=99bcdd5a60281110a93317d09a11b9f4831f9701 > > in the non working case the interface is recognised by "lsusb", it is seen by "aconnect -l" but not by "aplay -l". > > I'm willing to help testing as much as I can. > Alberto See the ticket for more details. There you noticed that Alberto retried with a vanilla kernel later and the problem happens there, too. [TLDR for the rest of this mail: I'm adding this report to the list of tracked Linux kernel regressions; the text you find below is based on a few templates paragraphs you might have encountered already in similar form.] BTW, let me use this mail to also add the report to the list of tracked regressions to ensure it's doesn't fall through the cracks: #regzbot introduced: v5.4.59..v5.4.233 https://bugzilla.kernel.org/show_bug.cgi?id=217084 #regzbot title: alsa: Roland VS-100 stopped working #regzbot ignore-activity This isn't a regression? This issue or a fix for it are already discussed somewhere else? It was fixed already? You want to clarify when the regression started to happen? Or point out I got the title or something else totally wrong? Then just reply and tell me -- ideally while also telling regzbot about it, as explained by the page listed in the footer of this mail. Developers: When fixing the issue, remember to add 'Link:' tags pointing to the report (e.g. the buzgzilla ticket and maybe this mail as well, if this thread sees some discussion). See page linked in footer for details. Ciao, Thorsten (wearing his 'the Linux kernel's regression tracker' hat) -- Everything you wanna know about Linux kernel regression tracking: https://linux-regtracking.leemhuis.info/about/#tldr If I did something stupid, please tell me, as explained on that page.

1 0

[PATCH] usb: gadget: u_audio: don't let userspace block driver unbind
by Alvin Šipraga 03 Mar '23

03 Mar '23

From: Alvin Šipraga <alsi(a)bang-olufsen.dk> In the unbind callback for f_uac1 and f_uac2, a call to snd_card_free() via g_audio_cleanup() will disconnect the card and then wait for all resources to be released, which happens when the refcount falls to zero. Since userspace can keep the refcount incremented by not closing the relevant file descriptor, the call to unbind may block indefinitely. This can cause a deadlock during reboot, as evidenced by the following blocked task observed on my machine: task:reboot state:D stack:0 pid:2827 ppid:569 flags:0x0000000c Call trace: __switch_to+0xc8/0x140 __schedule+0x2f0/0x7c0 schedule+0x60/0xd0 schedule_timeout+0x180/0x1d4 wait_for_completion+0x78/0x180 snd_card_free+0x90/0xa0 g_audio_cleanup+0x2c/0x64 afunc_unbind+0x28/0x60 ... kernel_restart+0x4c/0xac __do_sys_reboot+0xcc/0x1ec __arm64_sys_reboot+0x28/0x30 invoke_syscall+0x4c/0x110 ... The issue can also be observed by opening the card with arecord and then stopping the process through the shell before unbinding: # arecord -D hw:UAC2Gadget -f S32_LE -c 2 -r 48000 /dev/null Recording WAVE '/dev/null' : Signed 32 bit Little Endian, Rate 48000 Hz, Stereo ^Z[1]+ Stopped arecord -D hw:UAC2Gadget -f S32_LE -c 2 -r 48000 /dev/null # echo gadget.0 > /sys/bus/gadget/drivers/configfs-gadget/unbind (observe that the unbind command never finishes) Fix the problem by using snd_card_free_when_closed() instead, which will still disconnect the card as desired, but defer the task of freeing the resources to the core once userspace closes its file descriptor. Fixes: 132fcb460839 ("usb: gadget: Add Audio Class 2.0 Driver") Cc: stable(a)vger.kernel.org Signed-off-by: Alvin Šipraga <alsi(a)bang-olufsen.dk> --- drivers/usb/gadget/function/u_audio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/usb/gadget/function/u_audio.c b/drivers/usb/gadget/function/u_audio.c index c1f62e91b012..4a42574b4a7f 100644 --- a/drivers/usb/gadget/function/u_audio.c +++ b/drivers/usb/gadget/function/u_audio.c @@ -1422,7 +1422,7 @@ void g_audio_cleanup(struct g_audio *g_audio) uac = g_audio->uac; card = uac->card; if (card) - snd_card_free(card); + snd_card_free_when_closed(card); kfree(uac->p_prm.reqs); kfree(uac->c_prm.reqs); -- 2.39.1

3 2

[PATCH 1/5] ASoC: cs35l45: Support for GPIO pins configuration.
by Vlad Karpovich 02 Mar '23

02 Mar '23

From: "Vlad.Karpovich" <vkarpovi(a)opensource.cirrus.com> Adds device tree configuration for cs35l45 GPIOs Signed-off-by: Vlad Karpovich <vkarpovi(a)opensource.cirrus.com> --- include/dt-bindings/sound/cs35l45.h | 57 +++++++++++++++++++++++++++++ sound/soc/codecs/cs35l45-tables.c | 14 +++++++ sound/soc/codecs/cs35l45.c | 56 ++++++++++++++++++++++++++++ sound/soc/codecs/cs35l45.h | 27 +++++++++++++- 4 files changed, 152 insertions(+), 2 deletions(-) diff --git a/include/dt-bindings/sound/cs35l45.h b/include/dt-bindings/sound/cs35l45.h index 076da4b2c28d..25386af18445 100644 --- a/include/dt-bindings/sound/cs35l45.h +++ b/include/dt-bindings/sound/cs35l45.h @@ -17,4 +17,61 @@ #define CS35L45_ASP_TX_HIZ_UNUSED 0x1 #define CS35L45_ASP_TX_HIZ_DISABLED 0x2 +/* + * Optional GPIOX Sub-nodes: + * The cs35l45 node can have up to three "cirrus,gpio-ctrlX" ('X' = [1,2,3]) + * sub-nodes for configuring the GPIO pins. + * + * - gpio-dir : GPIO pin direction. Valid only when 'gpio-ctrl' + * is 1. + * 0 = Output + * 1 = Input (Default) + * + * - gpio-lvl : GPIO level. Valid only when 'gpio-ctrl' is 1 and 'gpio-dir' is 0. + * + * 0 = Low (Default) + * 1 = High + * + * - gpio-op-cfg : GPIO output configuration. Valid only when 'gpio-ctrl' is 1 + * and 'gpio-dir' is 0. + * + * 0 = CMOS (Default) + * 1 = Open Drain + * + * - gpio-pol : GPIO output polarity select. Valid only when 'gpio-ctrl' is 1 + * and 'gpio-dir' is 0. + * + * 0 = Non-inverted, Active High (Default) + * 1 = Inverted, Active Low + * + * - gpio-invert : Defines the polarity of the GPIO pin if configured + * as input. + * + * 0 = Not inverted (Default) + * 1 = Inverted + * + * - gpio-ctrl : Defines the function of the GPIO pin. + * + * GPIO1: + * 0 = High impedance input (Default) + * 1 = Pin acts as a GPIO, direction controlled by 'gpio-dir' + * 2 = Pin acts as MDSYNC, direction controlled by MDSYNC + * 3-7 = Reserved + * + * GPIO2: + * 0 = High impedance input (Default) + * 1 = Pin acts as a GPIO, direction controlled by 'gpio-dir' + * 2 = Pin acts as open drain INT + * 3 = Reserved + * 4 = Pin acts as push-pull output INT. Active low. + * 5 = Pin acts as push-pull output INT. Active high. + * 6,7 = Reserved + * + * GPIO3: + * 0 = High impedance input (Default) + * 1 = Pin acts as a GPIO, direction controlled by 'gpio-dir' + * 2-7 = Reserved + */ +#define CS35L45_NUM_GPIOS 0x3 + #endif /* DT_CS35L45_H */ diff --git a/sound/soc/codecs/cs35l45-tables.c b/sound/soc/codecs/cs35l45-tables.c index 5a2c2e684ef9..bca7e830821f 100644 --- a/sound/soc/codecs/cs35l45-tables.c +++ b/sound/soc/codecs/cs35l45-tables.c @@ -43,6 +43,9 @@ EXPORT_SYMBOL_NS_GPL(cs35l45_apply_patch, SND_SOC_CS35L45_TABLES); static const struct reg_default cs35l45_defaults[] = { { CS35L45_BLOCK_ENABLES, 0x00003323 }, { CS35L45_BLOCK_ENABLES2, 0x00000010 }, + { CS35L45_SYNC_GPIO1, 0x00000007 }, + { CS35L45_INTB_GPIO2_MCLK_REF, 0x00000005 }, + { CS35L45_GPIO3, 0x00000005 }, { CS35L45_REFCLK_INPUT, 0x00000510 }, { CS35L45_GLOBAL_SAMPLE_RATE, 0x00000003 }, { CS35L45_ASP_ENABLES1, 0x00000000 }, @@ -61,6 +64,9 @@ static const struct reg_default cs35l45_defaults[] = { { CS35L45_ASPTX4_INPUT, 0x00000028 }, { CS35L45_ASPTX5_INPUT, 0x00000048 }, { CS35L45_AMP_PCM_CONTROL, 0x00100000 }, + { CS35L45_GPIO1_CTRL1, 0x81000001 }, + { CS35L45_GPIO2_CTRL1, 0x81000001 }, + { CS35L45_GPIO3_CTRL1, 0x81000001 }, }; static bool cs35l45_readable_reg(struct device *dev, unsigned int reg) @@ -72,6 +78,9 @@ static bool cs35l45_readable_reg(struct device *dev, unsigned int reg) case CS35L45_BLOCK_ENABLES: case CS35L45_BLOCK_ENABLES2: case CS35L45_ERROR_RELEASE: + case CS35L45_SYNC_GPIO1: + case CS35L45_INTB_GPIO2_MCLK_REF: + case CS35L45_GPIO3: case CS35L45_REFCLK_INPUT: case CS35L45_GLOBAL_SAMPLE_RATE: case CS35L45_ASP_ENABLES1: @@ -92,6 +101,10 @@ static bool cs35l45_readable_reg(struct device *dev, unsigned int reg) case CS35L45_AMP_PCM_CONTROL: case CS35L45_AMP_PCM_HPF_TST: case CS35L45_IRQ1_EINT_4: + case CS35L45_GPIO_STATUS1: + case CS35L45_GPIO1_CTRL1: + case CS35L45_GPIO2_CTRL1: + case CS35L45_GPIO3_CTRL1: return true; default: return false; @@ -107,6 +120,7 @@ static bool cs35l45_volatile_reg(struct device *dev, unsigned int reg) case CS35L45_ERROR_RELEASE: case CS35L45_AMP_PCM_HPF_TST: /* not cachable */ case CS35L45_IRQ1_EINT_4: + case CS35L45_GPIO_STATUS1: return true; default: return false; diff --git a/sound/soc/codecs/cs35l45.c b/sound/soc/codecs/cs35l45.c index d15b3b77c7eb..901f3647fbda 100644 --- a/sound/soc/codecs/cs35l45.c +++ b/sound/soc/codecs/cs35l45.c @@ -536,7 +536,63 @@ static int __maybe_unused cs35l45_runtime_resume(struct device *dev) static int cs35l45_apply_property_config(struct cs35l45_private *cs35l45) { + struct device_node *node = cs35l45->dev->of_node; + unsigned int gpio_regs[] = {CS35L45_GPIO1_CTRL1, CS35L45_GPIO2_CTRL1, + CS35L45_GPIO3_CTRL1}; + unsigned int pad_regs[] = {CS35L45_SYNC_GPIO1, + CS35L45_INTB_GPIO2_MCLK_REF, CS35L45_GPIO3}; + struct device_node *child; unsigned int val; + char of_name[32]; + int ret, i; + + if (!node) + return 0; + + for (i = 0; i < CS35L45_NUM_GPIOS; i++) { + sprintf(of_name, "cirrus,gpio-ctrl%d", i + 1); + child = of_get_child_by_name(node, of_name); + if (!child) + continue; + + ret = of_property_read_u32(child, "gpio-dir", &val); + if (!ret) + regmap_update_bits(cs35l45->regmap, gpio_regs[i], + CS35L45_GPIO_DIR_MASK, + val << CS35L45_GPIO_DIR_SHIFT); + + ret = of_property_read_u32(child, "gpio-lvl", &val); + if (!ret) + regmap_update_bits(cs35l45->regmap, gpio_regs[i], + CS35L45_GPIO_LVL_MASK, + val << CS35L45_GPIO_LVL_SHIFT); + + ret = of_property_read_u32(child, "gpio-op-cfg", &val); + if (!ret) + regmap_update_bits(cs35l45->regmap, gpio_regs[i], + CS35L45_GPIO_OP_CFG_MASK, + val << CS35L45_GPIO_OP_CFG_SHIFT); + + ret = of_property_read_u32(child, "gpio-pol", &val); + if (!ret) + regmap_update_bits(cs35l45->regmap, gpio_regs[i], + CS35L45_GPIO_POL_MASK, + val << CS35L45_GPIO_POL_SHIFT); + + ret = of_property_read_u32(child, "gpio-ctrl", &val); + if (!ret) + regmap_update_bits(cs35l45->regmap, pad_regs[i], + CS35L45_GPIO_CTRL_MASK, + val << CS35L45_GPIO_CTRL_SHIFT); + + ret = of_property_read_u32(child, "gpio-invert", &val); + if (!ret) + regmap_update_bits(cs35l45->regmap, pad_regs[i], + CS35L45_GPIO_INVERT_MASK, + val << CS35L45_GPIO_INVERT_SHIFT); + + of_node_put(child); + } if (device_property_read_u32(cs35l45->dev, "cirrus,asp-sdout-hiz-ctrl", &val) == 0) { diff --git a/sound/soc/codecs/cs35l45.h b/sound/soc/codecs/cs35l45.h index 53fe9d2b7b15..f3a54fc57d53 100644 --- a/sound/soc/codecs/cs35l45.h +++ b/sound/soc/codecs/cs35l45.h @@ -14,6 +14,7 @@ #include <linux/pm_runtime.h> #include <linux/regmap.h> #include <linux/regulator/consumer.h> +#include <dt-bindings/sound/cs35l45.h> #define CS35L45_DEVID 0x00000000 #define CS35L45_REVID 0x00000004 @@ -24,6 +25,9 @@ #define CS35L45_BLOCK_ENABLES 0x00002018 #define CS35L45_BLOCK_ENABLES2 0x0000201C #define CS35L45_ERROR_RELEASE 0x00002034 +#define CS35L45_SYNC_GPIO1 0x00002430 +#define CS35L45_INTB_GPIO2_MCLK_REF 0x00002434 +#define CS35L45_GPIO3 0x00002438 #define CS35L45_REFCLK_INPUT 0x00002C04 #define CS35L45_GLOBAL_SAMPLE_RATE 0x00002C0C #define CS35L45_BOOST_CCM_CFG 0x00003808 @@ -48,8 +52,11 @@ #define CS35L45_AMP_PCM_CONTROL 0x00007000 #define CS35L45_AMP_PCM_HPF_TST 0x00007004 #define CS35L45_IRQ1_EINT_4 0x0000E01C -#define CS35L45_LASTREG 0x0000E01C - +#define CS35L45_GPIO_STATUS1 0x0000F000 +#define CS35L45_GPIO1_CTRL1 0x0000F008 +#define CS35L45_GPIO2_CTRL1 0x0000F00C +#define CS35L45_GPIO3_CTRL1 0x0000F010 +#define CS35L45_LASTREG 0x0000F010 /* SFT_RESET */ #define CS35L45_SOFT_RESET_TRIGGER 0x5A000000 @@ -165,6 +172,22 @@ #define CS35L45_OTP_BOOT_DONE_STS_MASK BIT(1) #define CS35L45_OTP_BUSY_MASK BIT(0) +/* GPIOX_CTRL1 */ +#define CS35L45_GPIO_DIR_SHIFT 31 +#define CS35L45_GPIO_DIR_MASK BIT(31) +#define CS35L45_GPIO_LVL_SHIFT 15 +#define CS35L45_GPIO_LVL_MASK BIT(15) +#define CS35L45_GPIO_OP_CFG_SHIFT 14 +#define CS35L45_GPIO_OP_CFG_MASK BIT(14) +#define CS35L45_GPIO_POL_SHIFT 12 +#define CS35L45_GPIO_POL_MASK BIT(12) + +/* SYNC_GPIO1, INTB_GPIO2_MCLK_REF, GPIO3 */ +#define CS35L45_GPIO_CTRL_SHIFT 20 +#define CS35L45_GPIO_CTRL_MASK GENMASK(22, 20) +#define CS35L45_GPIO_INVERT_SHIFT 19 +#define CS35L45_GPIO_INVERT_MASK BIT(19) + /* Mixer sources */ #define CS35L45_PCM_SRC_MASK 0x7F #define CS35L45_PCM_SRC_ZERO 0x00 -- 2.25.1

2 8

ALC4080 on MSI PRO X670-P WIFI
by GitHub issues - edited 02 Mar '23

02 Mar '23

alsa-project/alsa-ucm-conf issue #285 was edited from Onyx47: Hello! I have the board from the title. Out of the box, the audio does not work at all under pipewire (it does, for some reason, under pulseaudio), however the following change to `alsa/ucm2/USB-Audio/USB-Audio.conf` gets the rear panel working: ``` Regex "USB((0414:a00e)|(0b05:(1996|1a(16|2[07])))|(0db0(005a|151f|1feb|419c|82c7|a073|a47c|b202|d6e7|3130)))" ``` (USB id of the card is `0db0:3130`, just added `3130` to the regex). No luck on the front panel however, tried following some discussions and figuring it out to make a complete patch on my own but it's way over my head :) Here is my `alsa-info` output: http://alsa-project.org/db/?f=22aed0d76705b354a59364e6e8e41a4fde20e4da Let me know if there is any other info I need to provide. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/285 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 5/5] ASoC: cs35l45: Hibernation support
by Vlad Karpovich 02 Mar '23

02 Mar '23

From: "Vlad.Karpovich" <vkarpovi(a)opensource.cirrus.com> Adds support for the amplifier hibernation controlled by DSP firmware. Signed-off-by: Vlad Karpovich <vkarpovi(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l45-i2c.c | 2 + sound/soc/codecs/cs35l45-spi.c | 1 + sound/soc/codecs/cs35l45-tables.c | 6 +++ sound/soc/codecs/cs35l45.c | 80 +++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l45.h | 29 +++++++++++ 5 files changed, 118 insertions(+) diff --git a/sound/soc/codecs/cs35l45-i2c.c b/sound/soc/codecs/cs35l45-i2c.c index 5224aaf584b8..1c644ddf66bc 100644 --- a/sound/soc/codecs/cs35l45-i2c.c +++ b/sound/soc/codecs/cs35l45-i2c.c @@ -33,6 +33,8 @@ static int cs35l45_i2c_probe(struct i2c_client *client) cs35l45->dev = dev; cs35l45->irq = client->irq; + cs35l45->bus_type = CONTROL_BUS_I2C; + cs35l45->i2c_addr = client->addr; return cs35l45_probe(cs35l45); } diff --git a/sound/soc/codecs/cs35l45-spi.c b/sound/soc/codecs/cs35l45-spi.c index bfb79255783e..3ea2e6fd2b88 100644 --- a/sound/soc/codecs/cs35l45-spi.c +++ b/sound/soc/codecs/cs35l45-spi.c @@ -36,6 +36,7 @@ static int cs35l45_spi_probe(struct spi_device *spi) cs35l45->dev = dev; cs35l45->irq = spi->irq; + cs35l45->bus_type = CONTROL_BUS_SPI; return cs35l45_probe(cs35l45); } diff --git a/sound/soc/codecs/cs35l45-tables.c b/sound/soc/codecs/cs35l45-tables.c index 6b68773979ec..212d9cb5a4fb 100644 --- a/sound/soc/codecs/cs35l45-tables.c +++ b/sound/soc/codecs/cs35l45-tables.c @@ -47,6 +47,8 @@ static const struct reg_default cs35l45_defaults[] = { { CS35L45_INTB_GPIO2_MCLK_REF, 0x00000005 }, { CS35L45_GPIO3, 0x00000005 }, { CS35L45_PWRMGT_CTL, 0x00000000 }, + { CS35L45_WAKESRC_CTL, 0x00000008 }, + { CS35L45_WKI2C_CTL, 0x00000030 }, { CS35L45_REFCLK_INPUT, 0x00000510 }, { CS35L45_GLOBAL_SAMPLE_RATE, 0x00000003 }, { CS35L45_ASP_ENABLES1, 0x00000000 }, @@ -126,6 +128,9 @@ static bool cs35l45_readable_reg(struct device *dev, unsigned int reg) case CS35L45_INTB_GPIO2_MCLK_REF: case CS35L45_GPIO3: case CS35L45_PWRMGT_CTL: + case CS35L45_WAKESRC_CTL: + case CS35L45_WKI2C_CTL: + case CS35L45_PWRMGT_STS: case CS35L45_REFCLK_INPUT: case CS35L45_GLOBAL_SAMPLE_RATE: case CS35L45_ASP_ENABLES1: @@ -210,6 +215,7 @@ static bool cs35l45_volatile_reg(struct device *dev, unsigned int reg) case CS35L45_GLOBAL_ENABLES: case CS35L45_ERROR_RELEASE: case CS35L45_AMP_PCM_HPF_TST: /* not cachable */ + case CS35L45_PWRMGT_STS: case CS35L45_IRQ1_STATUS: case CS35L45_IRQ1_EINT_1 ... CS35L45_IRQ1_EINT_18: case CS35L45_IRQ1_STS_1 ... CS35L45_IRQ1_STS_18: diff --git a/sound/soc/codecs/cs35l45.c b/sound/soc/codecs/cs35l45.c index c5de2c8a9fe3..a2bab613daa2 100644 --- a/sound/soc/codecs/cs35l45.c +++ b/sound/soc/codecs/cs35l45.c @@ -36,6 +36,8 @@ static bool cs35l45_check_cspl_mbox_sts(const enum cs35l45_cspl_mboxcmd cmd, return (sts == CSPL_MBOX_STS_RUNNING); case CSPL_MBOX_CMD_STOP_PRE_REINIT: return (sts == CSPL_MBOX_STS_RDY_FOR_REINIT); + case CSPL_MBOX_CMD_HIBERNATE: + return (sts == CSPL_MBOX_STS_HIBERNATE); default: return false; } @@ -745,11 +747,81 @@ static const struct snd_soc_component_driver cs35l45_component = { .endianness = 1, }; +static void cs35l45_setup_hibernate(struct cs35l45_private *cs35l45) +{ + unsigned int wksrc; + + if (cs35l45->bus_type == CONTROL_BUS_I2C) + wksrc = CS35L45_WKSRC_I2C; + else + wksrc = CS35L45_WKSRC_SPI; + + regmap_update_bits(cs35l45->regmap, CS35L45_WAKESRC_CTL, + CS35L45_WKSRC_EN_MASK, + wksrc << CS35L45_WKSRC_EN_SHIFT); + + regmap_set_bits(cs35l45->regmap, CS35L45_WAKESRC_CTL, + CS35L45_UPDT_WKCTL_MASK); + + regmap_update_bits(cs35l45->regmap, CS35L45_WKI2C_CTL, + CS35L45_WKI2C_ADDR_MASK, cs35l45->i2c_addr); + + regmap_set_bits(cs35l45->regmap, CS35L45_WKI2C_CTL, + CS35L45_UPDT_WKI2C_MASK); +} + +static int cs35l45_enter_hibernate(struct cs35l45_private *cs35l45) +{ + dev_dbg(cs35l45->dev, "Enter hibernate\n"); + + cs35l45_setup_hibernate(cs35l45); + + // Don't wait for ACK since bus activity would wake the device + regmap_write(cs35l45->regmap, CS35L45_DSP_VIRT1_MBOX_1, CSPL_MBOX_CMD_HIBERNATE); + + return 0; +} + +static int cs35l45_exit_hibernate(struct cs35l45_private *cs35l45) +{ + const int wake_retries = 20; + const int sleep_retries = 5; + int ret, i, j; + + for (i = 0; i < sleep_retries; i++) { + dev_dbg(cs35l45->dev, "Exit hibernate\n"); + + for (j = 0; j < wake_retries; j++) { + ret = cs35l45_set_cspl_mbox_cmd(cs35l45, cs35l45->regmap, + CSPL_MBOX_CMD_OUT_OF_HIBERNATE); + if (!ret) { + dev_dbg(cs35l45->dev, "Wake success at cycle: %d\n", j); + return 0; + } + usleep_range(100, 200); + } + + dev_err(cs35l45->dev, "Wake failed, re-enter hibernate: %d\n", ret); + + cs35l45_setup_hibernate(cs35l45); + } + + dev_err(cs35l45->dev, "Timed out waking device\n"); + + return -ETIMEDOUT; +} + static int __maybe_unused cs35l45_runtime_suspend(struct device *dev) { struct cs35l45_private *cs35l45 = dev_get_drvdata(dev); + if (!cs35l45->dsp.preloaded || !cs35l45->dsp.cs_dsp.running) + return 0; + + cs35l45_enter_hibernate(cs35l45); + regcache_cache_only(cs35l45->regmap, true); + regcache_mark_dirty(cs35l45->regmap); dev_dbg(cs35l45->dev, "Runtime suspended\n"); @@ -761,9 +833,17 @@ static int __maybe_unused cs35l45_runtime_resume(struct device *dev) struct cs35l45_private *cs35l45 = dev_get_drvdata(dev); int ret; + if (!cs35l45->dsp.preloaded || !cs35l45->dsp.cs_dsp.running) + return 0; + dev_dbg(cs35l45->dev, "Runtime resume\n"); regcache_cache_only(cs35l45->regmap, false); + + ret = cs35l45_exit_hibernate(cs35l45); + if (ret) + return ret; + ret = regcache_sync(cs35l45->regmap); if (ret != 0) dev_warn(cs35l45->dev, "regcache_sync failed: %d\n", ret); diff --git a/sound/soc/codecs/cs35l45.h b/sound/soc/codecs/cs35l45.h index 87032619b341..0da28439f628 100644 --- a/sound/soc/codecs/cs35l45.h +++ b/sound/soc/codecs/cs35l45.h @@ -30,6 +30,9 @@ #define CS35L45_INTB_GPIO2_MCLK_REF 0x00002434 #define CS35L45_GPIO3 0x00002438 #define CS35L45_PWRMGT_CTL 0x00002900 +#define CS35L45_WAKESRC_CTL 0x00002904 +#define CS35L45_WKI2C_CTL 0x00002908 +#define CS35L45_PWRMGT_STS 0x0000290C #define CS35L45_REFCLK_INPUT 0x00002C04 #define CS35L45_GLOBAL_SAMPLE_RATE 0x00002C0C #define CS35L45_BOOST_CCM_CFG 0x00003808 @@ -348,6 +351,25 @@ #define CS35L45_POST_GLOBAL_EN_US 5000 #define CS35L45_PRE_GLOBAL_DIS_US 3000 +/* WAKESRC_CTL */ +#define CS35L45_WKSRC_SYNC_GPIO1 BIT(0) +#define CS35L45_WKSRC_INT_GPIO2 BIT(1) +#define CS35L45_WKSRC_GPIO3 BIT(2) +#define CS35L45_WKSRC_SPI BIT(3) +#define CS35L45_WKSRC_I2C BIT(4) +#define CS35L45_UPDT_WKCTL_SHIFT 15 +#define CS35L45_UPDT_WKCTL_MASK BIT(15) +#define CS35L45_WKSRC_EN_SHIFT 8 +#define CS35L45_WKSRC_EN_MASK GENMASK(12, 8) +#define CS35L45_WKSRC_POL_SHIFT 0 +#define CS35L45_WKSRC_POL_MASK GENMASK(3, 0) + +/* WAKEI2C_CTL */ +#define CS35L45_UPDT_WKI2C_SHIFT 15 +#define CS35L45_UPDT_WKI2C_MASK BIT(15) +#define CS35L45_WKI2C_ADDR_SHIFT 0 +#define CS35L45_WKI2C_ADDR_MASK GENMASK(6, 0) + #define CS35L45_SPI_MAX_FREQ 4000000 enum cs35l45_cspl_mboxstate { @@ -369,6 +391,11 @@ enum cs35l45_cspl_mboxcmd { CSPL_MBOX_CMD_INVALID_SEQUENCE = -2, }; +enum control_bus_type { + CONTROL_BUS_I2C = 0, + CONTROL_BUS_SPI = 1, +}; + #define CS35L45_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_3LE| \ SNDRV_PCM_FMTBIT_S24_LE) @@ -439,6 +466,8 @@ struct cs35l45_private { u8 slot_count; int irq_invert; int irq; + unsigned int i2c_addr; + enum control_bus_type bus_type; struct regmap_irq_chip_data *irq_data; }; -- 2.25.1

2 1

[PATCH 5/5] ASoC: cs34l45: Hibernation support
by Vlad Karpovich 02 Mar '23

02 Mar '23

From: "Vlad.Karpovich" <vkarpovi(a)opensource.cirrus.com> Adds support for the amplifier hibernation controlled by DSP firmware. Signed-off-by: Vlad Karpovich <vkarpovi(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l45-i2c.c | 2 + sound/soc/codecs/cs35l45-spi.c | 1 + sound/soc/codecs/cs35l45-tables.c | 6 +++ sound/soc/codecs/cs35l45.c | 80 +++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l45.h | 29 +++++++++++ 5 files changed, 118 insertions(+) diff --git a/sound/soc/codecs/cs35l45-i2c.c b/sound/soc/codecs/cs35l45-i2c.c index 5224aaf584b8..1c644ddf66bc 100644 --- a/sound/soc/codecs/cs35l45-i2c.c +++ b/sound/soc/codecs/cs35l45-i2c.c @@ -33,6 +33,8 @@ static int cs35l45_i2c_probe(struct i2c_client *client) cs35l45->dev = dev; cs35l45->irq = client->irq; + cs35l45->bus_type = CONTROL_BUS_I2C; + cs35l45->i2c_addr = client->addr; return cs35l45_probe(cs35l45); } diff --git a/sound/soc/codecs/cs35l45-spi.c b/sound/soc/codecs/cs35l45-spi.c index bfb79255783e..3ea2e6fd2b88 100644 --- a/sound/soc/codecs/cs35l45-spi.c +++ b/sound/soc/codecs/cs35l45-spi.c @@ -36,6 +36,7 @@ static int cs35l45_spi_probe(struct spi_device *spi) cs35l45->dev = dev; cs35l45->irq = spi->irq; + cs35l45->bus_type = CONTROL_BUS_SPI; return cs35l45_probe(cs35l45); } diff --git a/sound/soc/codecs/cs35l45-tables.c b/sound/soc/codecs/cs35l45-tables.c index 6b68773979ec..212d9cb5a4fb 100644 --- a/sound/soc/codecs/cs35l45-tables.c +++ b/sound/soc/codecs/cs35l45-tables.c @@ -47,6 +47,8 @@ static const struct reg_default cs35l45_defaults[] = { { CS35L45_INTB_GPIO2_MCLK_REF, 0x00000005 }, { CS35L45_GPIO3, 0x00000005 }, { CS35L45_PWRMGT_CTL, 0x00000000 }, + { CS35L45_WAKESRC_CTL, 0x00000008 }, + { CS35L45_WKI2C_CTL, 0x00000030 }, { CS35L45_REFCLK_INPUT, 0x00000510 }, { CS35L45_GLOBAL_SAMPLE_RATE, 0x00000003 }, { CS35L45_ASP_ENABLES1, 0x00000000 }, @@ -126,6 +128,9 @@ static bool cs35l45_readable_reg(struct device *dev, unsigned int reg) case CS35L45_INTB_GPIO2_MCLK_REF: case CS35L45_GPIO3: case CS35L45_PWRMGT_CTL: + case CS35L45_WAKESRC_CTL: + case CS35L45_WKI2C_CTL: + case CS35L45_PWRMGT_STS: case CS35L45_REFCLK_INPUT: case CS35L45_GLOBAL_SAMPLE_RATE: case CS35L45_ASP_ENABLES1: @@ -210,6 +215,7 @@ static bool cs35l45_volatile_reg(struct device *dev, unsigned int reg) case CS35L45_GLOBAL_ENABLES: case CS35L45_ERROR_RELEASE: case CS35L45_AMP_PCM_HPF_TST: /* not cachable */ + case CS35L45_PWRMGT_STS: case CS35L45_IRQ1_STATUS: case CS35L45_IRQ1_EINT_1 ... CS35L45_IRQ1_EINT_18: case CS35L45_IRQ1_STS_1 ... CS35L45_IRQ1_STS_18: diff --git a/sound/soc/codecs/cs35l45.c b/sound/soc/codecs/cs35l45.c index c5de2c8a9fe3..a2bab613daa2 100644 --- a/sound/soc/codecs/cs35l45.c +++ b/sound/soc/codecs/cs35l45.c @@ -36,6 +36,8 @@ static bool cs35l45_check_cspl_mbox_sts(const enum cs35l45_cspl_mboxcmd cmd, return (sts == CSPL_MBOX_STS_RUNNING); case CSPL_MBOX_CMD_STOP_PRE_REINIT: return (sts == CSPL_MBOX_STS_RDY_FOR_REINIT); + case CSPL_MBOX_CMD_HIBERNATE: + return (sts == CSPL_MBOX_STS_HIBERNATE); default: return false; } @@ -745,11 +747,81 @@ static const struct snd_soc_component_driver cs35l45_component = { .endianness = 1, }; +static void cs35l45_setup_hibernate(struct cs35l45_private *cs35l45) +{ + unsigned int wksrc; + + if (cs35l45->bus_type == CONTROL_BUS_I2C) + wksrc = CS35L45_WKSRC_I2C; + else + wksrc = CS35L45_WKSRC_SPI; + + regmap_update_bits(cs35l45->regmap, CS35L45_WAKESRC_CTL, + CS35L45_WKSRC_EN_MASK, + wksrc << CS35L45_WKSRC_EN_SHIFT); + + regmap_set_bits(cs35l45->regmap, CS35L45_WAKESRC_CTL, + CS35L45_UPDT_WKCTL_MASK); + + regmap_update_bits(cs35l45->regmap, CS35L45_WKI2C_CTL, + CS35L45_WKI2C_ADDR_MASK, cs35l45->i2c_addr); + + regmap_set_bits(cs35l45->regmap, CS35L45_WKI2C_CTL, + CS35L45_UPDT_WKI2C_MASK); +} + +static int cs35l45_enter_hibernate(struct cs35l45_private *cs35l45) +{ + dev_dbg(cs35l45->dev, "Enter hibernate\n"); + + cs35l45_setup_hibernate(cs35l45); + + // Don't wait for ACK since bus activity would wake the device + regmap_write(cs35l45->regmap, CS35L45_DSP_VIRT1_MBOX_1, CSPL_MBOX_CMD_HIBERNATE); + + return 0; +} + +static int cs35l45_exit_hibernate(struct cs35l45_private *cs35l45) +{ + const int wake_retries = 20; + const int sleep_retries = 5; + int ret, i, j; + + for (i = 0; i < sleep_retries; i++) { + dev_dbg(cs35l45->dev, "Exit hibernate\n"); + + for (j = 0; j < wake_retries; j++) { + ret = cs35l45_set_cspl_mbox_cmd(cs35l45, cs35l45->regmap, + CSPL_MBOX_CMD_OUT_OF_HIBERNATE); + if (!ret) { + dev_dbg(cs35l45->dev, "Wake success at cycle: %d\n", j); + return 0; + } + usleep_range(100, 200); + } + + dev_err(cs35l45->dev, "Wake failed, re-enter hibernate: %d\n", ret); + + cs35l45_setup_hibernate(cs35l45); + } + + dev_err(cs35l45->dev, "Timed out waking device\n"); + + return -ETIMEDOUT; +} + static int __maybe_unused cs35l45_runtime_suspend(struct device *dev) { struct cs35l45_private *cs35l45 = dev_get_drvdata(dev); + if (!cs35l45->dsp.preloaded || !cs35l45->dsp.cs_dsp.running) + return 0; + + cs35l45_enter_hibernate(cs35l45); + regcache_cache_only(cs35l45->regmap, true); + regcache_mark_dirty(cs35l45->regmap); dev_dbg(cs35l45->dev, "Runtime suspended\n"); @@ -761,9 +833,17 @@ static int __maybe_unused cs35l45_runtime_resume(struct device *dev) struct cs35l45_private *cs35l45 = dev_get_drvdata(dev); int ret; + if (!cs35l45->dsp.preloaded || !cs35l45->dsp.cs_dsp.running) + return 0; + dev_dbg(cs35l45->dev, "Runtime resume\n"); regcache_cache_only(cs35l45->regmap, false); + + ret = cs35l45_exit_hibernate(cs35l45); + if (ret) + return ret; + ret = regcache_sync(cs35l45->regmap); if (ret != 0) dev_warn(cs35l45->dev, "regcache_sync failed: %d\n", ret); diff --git a/sound/soc/codecs/cs35l45.h b/sound/soc/codecs/cs35l45.h index 87032619b341..0da28439f628 100644 --- a/sound/soc/codecs/cs35l45.h +++ b/sound/soc/codecs/cs35l45.h @@ -30,6 +30,9 @@ #define CS35L45_INTB_GPIO2_MCLK_REF 0x00002434 #define CS35L45_GPIO3 0x00002438 #define CS35L45_PWRMGT_CTL 0x00002900 +#define CS35L45_WAKESRC_CTL 0x00002904 +#define CS35L45_WKI2C_CTL 0x00002908 +#define CS35L45_PWRMGT_STS 0x0000290C #define CS35L45_REFCLK_INPUT 0x00002C04 #define CS35L45_GLOBAL_SAMPLE_RATE 0x00002C0C #define CS35L45_BOOST_CCM_CFG 0x00003808 @@ -348,6 +351,25 @@ #define CS35L45_POST_GLOBAL_EN_US 5000 #define CS35L45_PRE_GLOBAL_DIS_US 3000 +/* WAKESRC_CTL */ +#define CS35L45_WKSRC_SYNC_GPIO1 BIT(0) +#define CS35L45_WKSRC_INT_GPIO2 BIT(1) +#define CS35L45_WKSRC_GPIO3 BIT(2) +#define CS35L45_WKSRC_SPI BIT(3) +#define CS35L45_WKSRC_I2C BIT(4) +#define CS35L45_UPDT_WKCTL_SHIFT 15 +#define CS35L45_UPDT_WKCTL_MASK BIT(15) +#define CS35L45_WKSRC_EN_SHIFT 8 +#define CS35L45_WKSRC_EN_MASK GENMASK(12, 8) +#define CS35L45_WKSRC_POL_SHIFT 0 +#define CS35L45_WKSRC_POL_MASK GENMASK(3, 0) + +/* WAKEI2C_CTL */ +#define CS35L45_UPDT_WKI2C_SHIFT 15 +#define CS35L45_UPDT_WKI2C_MASK BIT(15) +#define CS35L45_WKI2C_ADDR_SHIFT 0 +#define CS35L45_WKI2C_ADDR_MASK GENMASK(6, 0) + #define CS35L45_SPI_MAX_FREQ 4000000 enum cs35l45_cspl_mboxstate { @@ -369,6 +391,11 @@ enum cs35l45_cspl_mboxcmd { CSPL_MBOX_CMD_INVALID_SEQUENCE = -2, }; +enum control_bus_type { + CONTROL_BUS_I2C = 0, + CONTROL_BUS_SPI = 1, +}; + #define CS35L45_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_3LE| \ SNDRV_PCM_FMTBIT_S24_LE) @@ -439,6 +466,8 @@ struct cs35l45_private { u8 slot_count; int irq_invert; int irq; + unsigned int i2c_addr; + enum control_bus_type bus_type; struct regmap_irq_chip_data *irq_data; }; -- 2.25.1

1 0

ALC4080 on MSI PRO X670-P WIFI
by GitHub issues - opened 02 Mar '23

02 Mar '23

alsa-project/alsa-ucm-conf issue #285 was opened from Onyx47: Hello! I have the board from the title. Out of the box, the audio does not work at all, however the following change to `alsa/ucm2/USB-Audio/USB-Audio.conf` gets the rear panel working: ``` Regex "USB((0414:a00e)|(0b05:(1996|1a(16|2[07])))|(0db0(005a|151f|1feb|419c|82c7|a073|a47c|b202|d6e7|3130)))" ``` (USB id of the card is `0db0:3130`, just added `3130` to the regex). No luck on the front panel however, tried following some discussions and figuring it out to make a complete patch on my own but it's way over my head :) Here is my `alsa-info` output: http://alsa-project.org/db/?f=22aed0d76705b354a59364e6e8e41a4fde20e4da Let me know if there is any other info I need to provide. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/285 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH V2] ucm2: Rockchip: rk817: Add ALSA UCM support
by Chris Morgan 02 Mar '23

02 Mar '23

From: Chris Morgan <macromorgan(a)hotmail.com> Add ALSA-UCM support for the Rockchip RK817 audio codec. This codec is typically configured either with an internal or external amplifier as reflected by the longname. This configuration has been tested on the Anbernic RG353P (rk817_ext) and the Odroid Go Advance (rk817_int). Changes from V1: - Use a constant to control if the output mux is set to HP or SPK. Signed-off-by: Chris Morgan <macromorgan(a)hotmail.com> --- ucm2/Rockchip/rk817-sound/HiFi.conf | 68 ++++++++++++++++++++++ ucm2/Rockchip/rk817-sound/rk817-sound.conf | 6 ++ ucm2/conf.d/simple-card/rk817_ext.conf | 1 + ucm2/conf.d/simple-card/rk817_int.conf | 1 + 4 files changed, 76 insertions(+) create mode 100644 ucm2/Rockchip/rk817-sound/HiFi.conf create mode 100644 ucm2/Rockchip/rk817-sound/rk817-sound.conf create mode 120000 ucm2/conf.d/simple-card/rk817_ext.conf create mode 120000 ucm2/conf.d/simple-card/rk817_int.conf diff --git a/ucm2/Rockchip/rk817-sound/HiFi.conf b/ucm2/Rockchip/rk817-sound/HiFi.conf new file mode 100644 index 0000000..8d90567 --- /dev/null +++ b/ucm2/Rockchip/rk817-sound/HiFi.conf @@ -0,0 +1,68 @@ +Define.pbk_mux "SPK" + +If.1 { + Condition { + Type ControlExists + Control "name='Internal Speakers Switch'" + } + + True { + Define.pbk_mux "HP" + SectionDevice."Speaker".EnableSequence [ + cset "name='Internal Speakers Switch' on" + ] + + SectionDevice."Speaker".DisableSequence [ + cset "name='Internal Speakers Switch' off" + ] + } +} + +SectionDevice."Speaker" { + Comment "Internal Speaker" + + EnableSequence [ + cset "name='Playback Mux' ${var:pbk_mux}" + ] + + Value { + PlaybackMixerElem "Master Playback Volume" + PlaybackPriority 100 + PlaybackPCM "hw:${CardId}" + } + + ConflictingDevice [ + "Headphones" + ] +} + +SectionDevice."Mic" { + Comment "Microphone" + + Value { + CapturePriority 100 + CapturePCM "hw:${CardId}" + CaptureMixerElem "Mic Capture Gain" + CaptureMasterElem "Master Capture Volume" + } +} + +SectionDevice."Headphones" { + Comment "Headphones" + + EnableSequence [ + cset "name='Playback Mux' HP" + ] + + Value { + PlaybackMixerElem "Master Playback Volume" + PlaybackPriority 200 + PlaybackPCM "hw:${CardId}" + JackControl "Headphones Jack" + JackHWMute "Speaker" + } + + ConflictingDevice [ + "Speaker" + ] +} diff --git a/ucm2/Rockchip/rk817-sound/rk817-sound.conf b/ucm2/Rockchip/rk817-sound/rk817-sound.conf new file mode 100644 index 0000000..90ec794 --- /dev/null +++ b/ucm2/Rockchip/rk817-sound/rk817-sound.conf @@ -0,0 +1,6 @@ +Syntax 4 + +SectionUseCase."HiFi" { + File "/Rockchip/rk817-sound/HiFi.conf" + Comment "Play HiFi quality music" +} diff --git a/ucm2/conf.d/simple-card/rk817_ext.conf b/ucm2/conf.d/simple-card/rk817_ext.conf new file mode 120000 index 0000000..6fa8db7 --- /dev/null +++ b/ucm2/conf.d/simple-card/rk817_ext.conf @@ -0,0 +1 @@ +../../Rockchip/rk817-sound/rk817-sound.conf \ No newline at end of file diff --git a/ucm2/conf.d/simple-card/rk817_int.conf b/ucm2/conf.d/simple-card/rk817_int.conf new file mode 120000 index 0000000..6fa8db7 --- /dev/null +++ b/ucm2/conf.d/simple-card/rk817_int.conf @@ -0,0 +1 @@ +../../Rockchip/rk817-sound/rk817-sound.conf \ No newline at end of file -- 2.34.1

2 1

X13s
by GitHub pull_request - opened 02 Mar '23

02 Mar '23

alsa-project/alsa-ucm-conf pull request #284 was opened from Srinivas-Kandagatla: This patchset adds support for X13s ucm which includes support for Speakers, 2 DMICs on Panel and Headset with Mic. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/284 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/284.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Question: is it possible to subscribe a port to system announcer ?
by GitHub issues - opened 01 Mar '23

01 Mar '23

alsa-project/alsa-python issue #7 was opened from Houston4444: Hi. I try to add ALSA support to [Patchance](https://github.com/Houston4444/Patchance). I 'd like to create a port capable to receive events as SEQ_EVENT_PORT_START or SEQ_EVENT_PORT_EXIT in order to get a complete graph. I see in patchage source code (C++) that it uses 'snd_seq_connect_from' function to enable this events. Is it possible in python with this lib ? Thanks. Issue URL : https://github.com/alsa-project/alsa-python/issues/7 Repository URL: https://github.com/alsa-project/alsa-python

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[PATCH v2 0/2] ASoC: mediatek: add missing initialization
by Trevor Wu 01 Mar '23

01 Mar '23

Fix dts parsing problem in mt8188 and mt8195 etdm dai drivers. Changes since v1: - split the patch into two commits and add Fixes tag. Trevor Wu (2): ASoC: mediatek: mt8188: add missing initialization ASoC: mediatek: mt8195: add missing initialization sound/soc/mediatek/mt8188/mt8188-dai-etdm.c | 3 +++ sound/soc/mediatek/mt8195/mt8195-dai-etdm.c | 3 +++ 2 files changed, 6 insertions(+) -- 2.18.0

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[PATCH] ASoC: mediatek: add missing initialization
by Trevor Wu 01 Mar '23

01 Mar '23

In etdm dai driver, dai_etdm_parse_of() function is used to parse dts properties to get parameters. There are two for-loops which are sepearately for all etdm and etdm input only cases. In etdm in only loop, dai_id is not initialized, so it keeps the value intiliazed in another loop. In the patch, add the missing initialization to fix the unexpected parsing problem. Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> --- sound/soc/mediatek/mt8188/mt8188-dai-etdm.c | 3 +++ sound/soc/mediatek/mt8195/mt8195-dai-etdm.c | 3 +++ 2 files changed, 6 insertions(+) diff --git a/sound/soc/mediatek/mt8188/mt8188-dai-etdm.c b/sound/soc/mediatek/mt8188/mt8188-dai-etdm.c index 1c53d4cb19bb..7a37752d4244 100644 --- a/sound/soc/mediatek/mt8188/mt8188-dai-etdm.c +++ b/sound/soc/mediatek/mt8188/mt8188-dai-etdm.c @@ -2498,6 +2498,9 @@ static void mt8188_dai_etdm_parse_of(struct mtk_base_afe *afe) /* etdm in only */ for (i = 0; i < 2; i++) { + dai_id = ETDM_TO_DAI_ID(i); + etdm_data = afe_priv->dai_priv[dai_id]; + snprintf(prop, sizeof(prop), "mediatek,%s-chn-disabled", of_afe_etdms[i].name); diff --git a/sound/soc/mediatek/mt8195/mt8195-dai-etdm.c b/sound/soc/mediatek/mt8195/mt8195-dai-etdm.c index c2e268054773..f2c9a1fdbe0d 100644 --- a/sound/soc/mediatek/mt8195/mt8195-dai-etdm.c +++ b/sound/soc/mediatek/mt8195/mt8195-dai-etdm.c @@ -2567,6 +2567,9 @@ static void mt8195_dai_etdm_parse_of(struct mtk_base_afe *afe) /* etdm in only */ for (i = 0; i < 2; i++) { + dai_id = ETDM_TO_DAI_ID(i); + etdm_data = afe_priv->dai_priv[dai_id]; + ret = snprintf(prop, sizeof(prop), "mediatek,%s-chn-disabled", of_afe_etdms[i].name); -- 2.18.0

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[Patch] sound/aoa: Add support for iBook G4 (powerbook6,5)
by Thaison Nguyen 01 Mar '23

01 Mar '23

The audio on my iBook G4 (powerbook65) has not been working. This is because the ids of the device were missing in the aoa audio driver. With this patch, the IDs of Powerbook6,4 and Powerbook6,5 are added to the driver. Signed-off-by: Thaison Nguyen <mail(a)thaison-nguyen.de> --- sound/aoa/fabrics/layout.c | 15 +++++++++++++++ sound/aoa/soundbus/i2sbus/core.c | 2 +- 2 files changed, 16 insertions(+), 1 deletions(-) diff a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -111,7 +111,9 @@ MODULE_ALIAS("aoa-device-id-14"); MODULE_ALIAS("aoa-device-id-22"); MODULE_ALIAS("aoa-device-id-31"); MODULE_ALIAS("aoa-device-id-35"); +MODULE_ALIAS("aoa-device-id-38"); +MODULE_ALIAS("aoa-device-id-40"); MODULE_ALIAS("aoa-device-id-44"); /* onyx with all but microphone connected */ @@ -368,7 +368,20 @@ .connections = tas_connections_nolineout, }, }, + /* PowerBook6,4 */ + { .device_id = 40, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_all, + }, + }, /* PowerBook6,5 */ + { .device_id = 38, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_all, + }, + }, { .device_id = 44, .codecs[0] = { .name = "tas", diff a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -197,7 +197,7 @@ * so restrict to those we do handle for now. */ if (id && (*id == 22 || *id == 14 || *id == 35 || - *id == 31 || *id == 44)) { + *id == 31 || *id == 44 || *id == 40 || *id == 38)) { snprintf(dev->sound.modalias, 32, "aoa-device-id-%d", *id); ok = 1;

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Re: Patch for audio on iBook G4 (powerbook6,5)
by Christophe Leroy 01 Mar '23

01 Mar '23

Le 28/02/2023 à 22:17, mail(a)thaison-nguyen.de a écrit : > Hi > > I have added a bug report and a patch regarding audio for iBook G4 > (powerbook6,5) on kernel.org bugzilla, but nobody has picked it up. > https://bugzilla.kernel.org/show_bug.cgi?id=216504 > > Now I am trying to submit the patch here. > Please read and follow the process to submit a patch : https://docs.kernel.org/process/5.Posting.html Patches must be in the core part of the mail and not as an attached file. Patches must have a description of the change. Patches must be signed Patches must be sent to the relevant maintainer(s) based on the modified files. As far as I can see your patch is for the sound subsystem. Christophe

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[PATCH v3 0/3] ASoC: mchp-pdmc: fix poc noises when starting capture
by Claudiu Beznea 28 Feb '23

28 Feb '23

To start capture on Microchip PDMC the enable bits for each supported microphone need to be set. After this bit is set the PDMC starts to receive data from microphones and it considers this data as valid data. Thus if microphones are not ready the PDMC captures anyway data from its lines. This data is interpreted by the human ear as poc noises. To avoid this the following software workaround need to be applied when starting capture: 1/ enable PDMC channel 2/ wait 150ms 3/ execute 16 dummy reads from RHR 4/ clear interrupts 5/ enable interrupts 6/ enable DMA channel For this workaround to work step 6 need to be executed at the end. For step 6 was added patch 1/3 from this series. With this, component DAI driver sets its struct snd_soc_component_driver::start_dma_last = 1 and proper action is taken based on this flag when starting DAI trigger vs DMA. Thank you, Claudiu Beznea Changes in v3: - update the commit message of patch 2/3 to be more descriptive - in patch 3/3 initialize dd->startup_delay_us before calling of_property_read_u32() and don't check its return value; property is optional and the the default value is 150ms Changes in v2: - patch 1/3 from v1 is now "ASoC: soc-pcm: add option to start DMA after DAI" - pass start_dma_last from component DAI driver object (struct snd_soc_component_driver::start_dma_last); adapt patch 3/3 after this; - in patch 1/3 s/Do we need to start dma first/Do we need to start dma last in comment from soc_pcm_trigger() - collect review tag from Krzysztof Claudiu Beznea (3): ASoC: soc-pcm: add option to start DMA after DAI ASoC: dt-bindings: sama7g5-pdmc: add microchip,startup-delay-us binding ASoC: mchp-pdmc: fix poc noise at capture startup Claudiu Beznea (3): ASoC: soc-pcm: add option to start DMA after DAI ASoC: dt-bindings: sama7g5-pdmc: add microchip,startup-delay-us binding ASoC: mchp-pdmc: fix poc noise at capture startup .../sound/microchip,sama7g5-pdmc.yaml | 6 +++ include/sound/soc-component.h | 2 + sound/soc/atmel/mchp-pdmc.c | 53 +++++++++++++++++-- sound/soc/soc-pcm.c | 27 ++++++++-- 4 files changed, 78 insertions(+), 10 deletions(-) -- 2.34.1

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[PATCH] ASoC: zl38060 add gpiolib dependency
by Arnd Bergmann 28 Feb '23

28 Feb '23

From: Arnd Bergmann <arnd(a)arndb.de> Without gpiolib, this driver fails to link: arm-linux-gnueabi-ld: sound/soc/codecs/zl38060.o: in function `chip_gpio_get': zl38060.c:(.text+0x30): undefined reference to `gpiochip_get_data' arm-linux-gnueabi-ld: sound/soc/codecs/zl38060.o: in function `zl38_spi_probe': zl38060.c:(.text+0xa18): undefined reference to `devm_gpiochip_add_data_with_key' This appears to have been in the driver since the start, but is hard to hit in randconfig testing since gpiolib is almost always selected by something else. Fixes: 52e8a94baf90 ("ASoC: Add initial ZL38060 driver") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bd72c426a93d..07747565c3b5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -2103,6 +2103,7 @@ config SND_SOC_WSA883X config SND_SOC_ZL38060 tristate "Microsemi ZL38060 Connected Home Audio Processor" depends on SPI_MASTER + depends on GPIOLIB select REGMAP help Support for ZL38060 Connected Home Audio Processor from Microsemi, -- 2.39.2

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[PATCH] ASoC: sam9g20ek: Disable capture unless building with microphone input
by Mark Brown 28 Feb '23

28 Feb '23

Without modification the AT91SAM9G20-EK has no capture support, none of the inputs of the CODEC are wired to anything to useful and there are no paths supporting loopback. Since the audio is clocked from the CODEC and the DAPM inputs are marked as unusable this means that capture will fail to transfer any data as the ADC path can't be powered up. Flag this in the device description so apps don't see unusable capture support, guarded with the existing optional define for mic input. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/atmel/sam9g20_wm8731.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 1430642c8433..785b9d01d8af 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -98,6 +98,9 @@ static struct snd_soc_dai_link at91sam9g20ek_dai = { .init = at91sam9g20ek_wm8731_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBP_CFP, +#ifndef ENABLE_MIC_INPUT + .playback_only = true, +#endif SND_SOC_DAILINK_REG(pcm), }; --- base-commit: cac85e4616b1cf4a90844b952b49b9cbc4562530 change-id: 20230225-asoc-sam9g20ek-f4e775205d8c Best regards, -- Mark Brown <broonie(a)kernel.org>

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[PATCH 0/2] ASoC: mt8183: Fixes from an initial glance at a kselftest run
by Mark Brown 28 Feb '23

28 Feb '23

This is a collection of fixes I came up after glancing through an initial test run with the snappily named Kukui Jacuzzi SKU16 Chromebook on KernelCI. There are more issues flagged, this is just what I fixed thus far. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Mark Brown (2): ASoC: mt8183: Remove spammy logging from I2S DAI driver ASoC: mt8183: Fix event generation for I2S DAI operations sound/soc/mediatek/mt8183/mt8183-dai-i2s.c | 21 +++------------------ 1 file changed, 3 insertions(+), 18 deletions(-) --- base-commit: d2980d8d826554fa6981d621e569a453787472f8 change-id: 20230224-asoc-mt8183-quick-fixes-ccb7c567c755 Best regards, -- Mark Brown <broonie(a)kernel.org>

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[PATCH 0/3] ASoC: mt6358: Fixes from an initial glance at a kselftest run
by Mark Brown 28 Feb '23

28 Feb '23

This is a collection of fixes I came up after glancing through an initial test run with the snappily named Kukui Jacuzzi SKU16 Chromebook on KernelCI. There are more issues flagged, this is just what I fixed thus far. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Mark Brown (3): ASoC: mt6358: Fix event generation for wake on voice stage 2 switch ASoC: mt6358: Validate Wake on Voice 2 writes ASoC: mt6358: Remove undefined HPx Mux enumeration values sound/soc/codecs/mt6358.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) --- base-commit: d2980d8d826554fa6981d621e569a453787472f8 change-id: 20230224-asoc-mt6358-quick-fixes-71b4ea6b8f13 Best regards, -- Mark Brown <broonie(a)kernel.org>

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[PATCH 0/4] ASoC: mt8192: Fixes from initial glance at kselftest run
by Mark Brown 28 Feb '23

28 Feb '23

This is a collection of fixes I came up after glancing through an initial test run with the Spherion Chromebook on KernelCI. There are more issues flagged, this is just what I fixed thus far - the volume controls on the MT6359 have issues for example, and a lot of controls aren't marked as Switches like they should be. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Mark Brown (4): ASoC: mt8192: Remove spammy log messages ASoC: mt8192: Fix event generation for controls ASoC: mt8192: Report an error if when an invalid sidetone gain is written ASoC: mt8192: Fix range for sidetone positive gain sound/soc/mediatek/mt8192/mt8192-dai-adda.c | 58 ++++++++++------------------- 1 file changed, 19 insertions(+), 39 deletions(-) --- base-commit: b361d5d2464a88184f6e17a6462719ba79180b1a change-id: 20230223-asoc-mt8192-quick-fixes-8f3e0a187226 Best regards, -- Mark Brown <broonie(a)kernel.org>

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[RFC PATCH v2 00/22] Introduce QC USB SND audio offloading support
by Wesley Cheng 28 Feb '23

28 Feb '23

Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. USB SND: - Added a platform ops during usb_audio_suspend(). This allows for the USB offload driver to halt the audio stream when system enters PM suspend. This ensures the audio DSP is not issuing transfers on the USB bus. - Do not override platform ops if they are already populated. - Introduce a shared status variable between the USB offload and USB SND layers, to ensure that only one path is active at a time. If the USB bus is occupied, then userspace is notified that the path is busy. Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver Adding ASoC binding layer: soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP: q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userpsace ALSA can take action accordingly. Introduce XHCI interrupter support: XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver: qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platorm card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Mathias Nyman (4): xhci: fix event ring segment table related masks and variables in header xhci: remove xhci_test_trb_in_td_math early development check xhci: Refactor interrupter code for initial multi interrupter support. xhci: Add support to allocate several interrupters Wesley Cheng (18): usb: xhci: Add XHCI APIs to support USB offloading usb: host: xhci-mem: Cleanup pending secondary event ring events ASoC: Add SOC USB APIs for adding an USB backend ASoC: dt-bindings: Add USB_RX port ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qdsp6: q6afe: Increase APR timeout ASoC: qcom: Add USB backend ASoC driver for Q6 sound: usb: card: Introduce USB SND platform op callbacks sound: usb: Export USB SND APIs for modules dt-bindings: usb: dwc3: Add snps,num-hc-interrupters definition usb: dwc3: Add DT parameter to specify maximum number of interrupters sound: usb: Introduce QC USB SND offloading support sound: usb: card: Check for support for requested audio format sound: soc: soc-usb: Add PCM format check API for USB backend sound: soc: qcom: qusb6: Ensure PCM format is supported by USB audio device sound: usb: Prevent starting of audio stream if in use ASoC: dt-bindings: Add Q6USB backend bindings ASoC: dt-bindings: Update example for enabling USB offload on SM8250 .../bindings/sound/qcom,q6usb-dais.yaml | 55 + .../bindings/sound/qcom,sm8250.yaml | 13 + .../devicetree/bindings/usb/snps,dwc3.yaml | 12 + drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 5 +- drivers/usb/host/xhci-debugfs.c | 2 +- drivers/usb/host/xhci-mem.c | 471 +++-- drivers/usb/host/xhci-ring.c | 68 +- drivers/usb/host/xhci.c | 199 +- drivers/usb/host/xhci.h | 81 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/usb/xhci-intr.h | 94 + include/sound/pcm_params.h | 4 + include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 36 + sound/core/oss/pcm_oss.c | 58 - sound/core/pcm_lib.c | 65 + sound/soc/Makefile | 2 +- sound/soc/qcom/Kconfig | 4 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 48 + sound/soc/qcom/qdsp6/q6afe.c | 185 +- sound/soc/qcom/qdsp6/q6afe.h | 46 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 8 + sound/soc/qcom/qdsp6/q6usb.c | 234 +++ sound/soc/soc-usb.c | 215 ++ sound/usb/Kconfig | 14 + sound/usb/Makefile | 2 +- sound/usb/card.c | 58 + sound/usb/card.h | 29 + sound/usb/endpoint.c | 2 + sound/usb/helper.c | 1 + sound/usb/pcm.c | 28 +- sound/usb/pcm.h | 12 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1789 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 41 files changed, 4546 insertions(+), 410 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6usb-dais.yaml create mode 100644 include/linux/usb/xhci-intr.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

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[PATCH V4 0/8] Add SoundWire support for AMD platforms
by Vijendar Mukunda 28 Feb '23

28 Feb '23

ACP IP(v6.x) block has two SoundWire manager instance support. This patchset adds support for AMD SoundWire manager driver. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> changes since v3: - add usleep_range() in command/response implementation. - add usleep_range() in clock stop sequence. - modify usleep_range() values throughout the code. - remove unncessary debug statement from code. - update comment in probe() call. - fix timeout condition checks in the code. - drop "ret" variable in amd_disable_sdw_manager(). Changes since v2: - Remove useless variable initializations. - Add helper function to interpret peripheral status. - Move runtime pm sequence to probe_work workqueue. - Use string "SoundWire" instead of "soundwire" in code. - Update comments in interrupt handler and probe sequence. - Rename "sdw_lock" as "acp_sdw_lock". - Remove __func__ from dev_dbg statements. Changes since v1: - Drop asoc tree based patches. will send asoc patches as a separate series. - Fixed double space errors. - Use dev instead of pci->dev. - Use SoundWire manager terminology. - Remove amd_sdw_compute_slave_ports() function and use exported sdw_compute_slave_ports() function. - Remove unused variable "num_ports" from amd_manager structure. - Drop startup and shutdown dai callbacks. - Drop reset_page_addr callback. - Use relative address offset to program SoundWire manager registers throughout the code. - Separate wake enable interrupt handling from slave status handling logic. - Use acp_mmio to program ACP common registers. - Use dai_runtime_array implementation in dai_ops. - Refactor port_ops callbacks. - Add comments in port_ops callbacks. - Add retry count logic in irq thread to address faulty case. - Add helper function to interpret command response. - Add generic bandwidth allocation dependency in Kconfig options. - Add comments for AMD SoundWire power modes. - Add missing timeout check in amd_init_sdw_manager callback. - Declare frameshape parameters in probe call. - Handle error case in clock stop sequence. - Add comments in pm_prepare and pm_ops callbacks. Vijendar Mukunda (8): soundwire: export sdw_compute_slave_ports() function soundwire: amd: Add support for AMD Manager driver soundwire: amd: register SoundWire manager dai ops soundwire: amd: enable build for AMD SoundWire manager driver soundwire: amd: add SoundWire manager interrupt handling soundwire: amd: add runtime pm ops for AMD SoundWire manager driver soundwire: amd: handle SoundWire wake enable interrupt soundwire: amd: add pm_prepare callback and pm ops support drivers/soundwire/Kconfig | 10 + drivers/soundwire/Makefile | 4 + drivers/soundwire/amd_manager.c | 1321 +++++++++++++++++ drivers/soundwire/amd_manager.h | 275 ++++ drivers/soundwire/bus.h | 9 + .../soundwire/generic_bandwidth_allocation.c | 12 +- include/linux/soundwire/sdw_amd.h | 109 ++ 7 files changed, 1731 insertions(+), 9 deletions(-) create mode 100644 drivers/soundwire/amd_manager.c create mode 100644 drivers/soundwire/amd_manager.h create mode 100644 include/linux/soundwire/sdw_amd.h -- 2.34.1

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[PATCH v2 0/3] Fixes and Improvements for CS35L41 HDA
by Stefan Binding 28 Feb '23

28 Feb '23

Fixes issue in calibration, where return codes were misinterpreted. Enable High Pass filter to reduce pops and clicks. Add improvement to ensure firmware and tuning files are always loaded together. This ensure the firmware is alsways running with valid coefficients. Stefan Binding (2): ALSA: hda: cs35l41: Ensure firmware/tuning pairs are always loaded ALSA: hda: cs35l41: Enable Amp High Pass Filter Vitaly Rodionov (1): ALSA: hda: cs35l41: Correct error condition handling sound/pci/hda/cs35l41_hda.c | 109 ++++++++++++++++----------------- sound/pci/hda/hda_cs_dsp_ctl.c | 4 +- 2 files changed, 56 insertions(+), 57 deletions(-) -- 2.34.1

3 6

[PATCH] soundwire: bus: Fix unbalanced pm_runtime_put() causing usage count underflow
by Richard Fitzgerald 27 Feb '23

27 Feb '23

This reverts commit 443a98e649b4 ("soundwire: bus: use pm_runtime_resume_and_get()") Change calls to pm_runtime_resume_and_get() back to pm_runtime_get_sync(). This fixes a usage count underrun caused by doing a pm_runtime_put() even though pm_runtime_resume_and_get() returned an error. The three affected functions ignore -EACCES error from trying to get pm_runtime, and carry on, including a put at the end of the function. But pm_runtime_resume_and_get() does not increment the usage count if it returns an error. So in the -EACCES case you must not call pm_runtime_put(). The documentation for pm_runtime_get_sync() says: "Consider using pm_runtime_resume_and_get() ... as this is likely to result in cleaner code." In this case I don't think it results in cleaner code because the pm_runtime_put() at the end of the function would have to be conditional on the return value from pm_runtime_resume_and_get() at the top of the function. pm_runtime_get_sync() doesn't have this problem because it always increments the count, so always needs a put. The code can just flow through and do the pm_runtime_put() unconditionally. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- drivers/soundwire/bus.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 76515c33e639..4fd221d0cc81 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -571,9 +571,11 @@ int sdw_nread(struct sdw_slave *slave, u32 addr, size_t count, u8 *val) { int ret; - ret = pm_runtime_resume_and_get(&slave->dev); - if (ret < 0 && ret != -EACCES) + ret = pm_runtime_get_sync(&slave->dev); + if (ret < 0 && ret != -EACCES) { + pm_runtime_put_noidle(&slave->dev); return ret; + } ret = sdw_nread_no_pm(slave, addr, count, val); @@ -595,9 +597,11 @@ int sdw_nwrite(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val) { int ret; - ret = pm_runtime_resume_and_get(&slave->dev); - if (ret < 0 && ret != -EACCES) + ret = pm_runtime_get_sync(&slave->dev); + if (ret < 0 && ret != -EACCES) { + pm_runtime_put_noidle(&slave->dev); return ret; + } ret = sdw_nwrite_no_pm(slave, addr, count, val); @@ -1565,9 +1569,10 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) sdw_modify_slave_status(slave, SDW_SLAVE_ALERT); - ret = pm_runtime_resume_and_get(&slave->dev); + ret = pm_runtime_get_sync(&slave->dev); if (ret < 0 && ret != -EACCES) { dev_err(&slave->dev, "Failed to resume device: %d\n", ret); + pm_runtime_put_noidle(&slave->dev); return ret; } -- 2.30.2

2 1

[PATCH] Fix snprintf format warnings during 'alsa' kselftest compilation
by Ivan Orlov 27 Feb '23

27 Feb '23

Fix 'alsa' kselftest compilation warnings by making snprintf format correspond the actual parameters types. Signed-off-by: Ivan Orlov <ivan.orlov0322(a)gmail.com> --- tools/testing/selftests/alsa/pcm-test.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/tools/testing/selftests/alsa/pcm-test.c b/tools/testing/selftests/alsa/pcm-test.c index f293c7d81009..52b109a162c6 100644 --- a/tools/testing/selftests/alsa/pcm-test.c +++ b/tools/testing/selftests/alsa/pcm-test.c @@ -318,7 +318,7 @@ static void test_pcm_time1(struct pcm_data *data, goto __close; } if (rchannels != channels) { - snprintf(msg, sizeof(msg), "channels unsupported %ld != %ld", channels, rchannels); + snprintf(msg, sizeof(msg), "channels unsupported %ld != %u", channels, rchannels); skip = true; goto __close; } @@ -329,7 +329,7 @@ static void test_pcm_time1(struct pcm_data *data, goto __close; } if (rrate != rate) { - snprintf(msg, sizeof(msg), "rate unsupported %ld != %ld", rate, rrate); + snprintf(msg, sizeof(msg), "rate unsupported %ld != %u", rate, rrate); skip = true; goto __close; } @@ -393,24 +393,24 @@ static void test_pcm_time1(struct pcm_data *data, frames = snd_pcm_writei(handle, samples, rate); if (frames < 0) { snprintf(msg, sizeof(msg), - "Write failed: expected %d, wrote %li", rate, frames); + "Write failed: expected %ld, wrote %li", rate, frames); goto __close; } if (frames < rate) { snprintf(msg, sizeof(msg), - "expected %d, wrote %li", rate, frames); + "expected %ld, wrote %li", rate, frames); goto __close; } } else { frames = snd_pcm_readi(handle, samples, rate); if (frames < 0) { snprintf(msg, sizeof(msg), - "expected %d, wrote %li", rate, frames); + "expected %ld, wrote %li", rate, frames); goto __close; } if (frames < rate) { snprintf(msg, sizeof(msg), - "expected %d, wrote %li", rate, frames); + "expected %ld, wrote %li", rate, frames); goto __close; } } -- 2.34.1

2 3

Re: [bug report] firewire: cdev: obsolete NULL check to detect IEC 61883-1 FCP region
by Takashi Sakamoto 27 Feb '23

27 Feb '23

Hi, (C.C.ed to LKML and alsa-devel) On Mon, Feb 27, 2023 at 02:06:51PM +0300, Dan Carpenter wrote: > Hello Takashi Sakamoto, > > The patch e699600232e0: "firewire: cdev: obsolete NULL check to > detect IEC 61883-1 FCP region" from Jan 20, 2023, leads to the > following Smatch static checker warning: > > drivers/firewire/core-transaction.c:947 handle_fcp_region_request() > warn: passing freed memory 'request' > > drivers/firewire/core-transaction.c > 930 fw_send_response(card, request, RCODE_TYPE_ERROR); > 931 > 932 return; > 933 } > 934 > 935 rcu_read_lock(); > 936 list_for_each_entry_rcu(handler, &address_handler_list, link) { > 937 if (is_enclosing_handler(handler, offset, request->length)) > 938 handler->address_callback(card, request, tcode, > ^^^^^^^ > This warning is because fwnet_receive_packet() has a kfree(r) on the > first return path. > > 939 destination, source, > 940 p->generation, offset, > 941 request->data, > 942 request->length, > 943 handler->callback_data); > 944 } > 945 rcu_read_unlock(); > 946 > --> 947 fw_send_response(card, request, RCODE_COMPLETE); > 948 } Thanks for your report. Fortunately, We can not see the access to the released memory since the fwnet's address handler is registered to high memory region (0x'0001'0000'0000 to 0x'ffff'e000'0000). The region does not overlap IEC 61883-1 FCP region (0x'ffff'f000'0b00 to 0x'ffff'f000'0f00). The handler is called from handle_exclusive_region_request() instead of handle_fcp_region_request(). However, the code in fwnet is against the design of address handler apparently. The callee never release the memory for the request structure directly. It should be done by the call of fw_send_response(). I'll correct it for next merge window; i.e. for v6.4. Thanks Takashi Sakamoto

1 0

[PATCH 1/4] ASoC: dt-bindings: wlf,wm8524: Convert to json-schema
by Chancel Liu 27 Feb '23

27 Feb '23

Convert the Wolfson WM8524 24-bit 192KHz Stereo DAC device tree binding documentation to json-schema. Signed-off-by: Chancel Liu <chancel.liu(a)nxp.com> --- .../devicetree/bindings/sound/wlf,wm8524.yaml | 37 +++++++++++++++++++ .../devicetree/bindings/sound/wm8524.txt | 16 -------- 2 files changed, 37 insertions(+), 16 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8524.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wm8524.txt diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8524.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8524.yaml new file mode 100644 index 000000000000..09c54cc7de95 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8524.yaml @@ -0,0 +1,37 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8524.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson WM8524 24-bit 192KHz Stereo DAC + +maintainers: + - patches(a)opensource.cirrus.com + +properties: + compatible: + const: wlf,wm8524 + + "#sound-dai-cells": + const: 0 + + wlf,mute-gpios: + maxItems: 1 + description: + a GPIO spec for the MUTE pin. + +required: + - compatible + - wlf,mute-gpios + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + wm8524: codec { + compatible = "wlf,wm8524"; + wlf,mute-gpios = <&gpio1 8 GPIO_ACTIVE_LOW>; + }; diff --git a/Documentation/devicetree/bindings/sound/wm8524.txt b/Documentation/devicetree/bindings/sound/wm8524.txt deleted file mode 100644 index f6c0c263b135..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8524.txt +++ /dev/null @@ -1,16 +0,0 @@ -WM8524 audio CODEC - -This device does not use I2C or SPI but a simple Hardware Control Interface. - -Required properties: - - - compatible : "wlf,wm8524" - - - wlf,mute-gpios: a GPIO spec for the MUTE pin. - -Example: - -wm8524: codec { - compatible = "wlf,wm8524"; - wlf,mute-gpios = <&gpio1 8 GPIO_ACTIVE_LOW>; -}; -- 2.25.1

3 15

[PATCH AUTOSEL 4.14 4/9] ASoC: kirkwood: Iterate over array indexes instead of using pointer math
by Sasha Levin 27 Feb '23

27 Feb '23

From: Kees Cook <keescook(a)chromium.org> [ Upstream commit b3bcedc0402fcdc5c8624c433562d9d1882749d8 ] Walking the dram->cs array was seen as accesses beyond the first array item by the compiler. Instead, use the array index directly. This allows for run-time bounds checking under CONFIG_UBSAN_BOUNDS as well. Seen with GCC 13 with -fstrict-flex-arrays: ../sound/soc/kirkwood/kirkwood-dma.c: In function 'kirkwood_dma_conf_mbus_windows.constprop': ../sound/soc/kirkwood/kirkwood-dma.c:90:24: warning: array subscript 0 is outside array bounds of 'const struct mbus_dram_window[0]' [-Warray-bounds=] 90 | if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { | ~~^~~~~~ Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> Link: https://lore.kernel.org/r/20230127224128.never.410-kees@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index 35ca8e8bb5e52..9736fb36082fb 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -90,7 +90,7 @@ kirkwood_dma_conf_mbus_windows(void __iomem *base, int win, /* try to find matching cs for current dma address */ for (i = 0; i < dram->num_cs; i++) { - const struct mbus_dram_window *cs = dram->cs + i; + const struct mbus_dram_window *cs = &dram->cs[i]; if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { writel(cs->base & 0xffff0000, base + KIRKWOOD_AUDIO_WIN_BASE_REG(win)); -- 2.39.0

1 0

[PATCH AUTOSEL 4.19 05/10] ASoC: kirkwood: Iterate over array indexes instead of using pointer math
by Sasha Levin 27 Feb '23

27 Feb '23

From: Kees Cook <keescook(a)chromium.org> [ Upstream commit b3bcedc0402fcdc5c8624c433562d9d1882749d8 ] Walking the dram->cs array was seen as accesses beyond the first array item by the compiler. Instead, use the array index directly. This allows for run-time bounds checking under CONFIG_UBSAN_BOUNDS as well. Seen with GCC 13 with -fstrict-flex-arrays: ../sound/soc/kirkwood/kirkwood-dma.c: In function 'kirkwood_dma_conf_mbus_windows.constprop': ../sound/soc/kirkwood/kirkwood-dma.c:90:24: warning: array subscript 0 is outside array bounds of 'const struct mbus_dram_window[0]' [-Warray-bounds=] 90 | if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { | ~~^~~~~~ Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> Link: https://lore.kernel.org/r/20230127224128.never.410-kees@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index 255cc45905b81..51f75523b691a 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -90,7 +90,7 @@ kirkwood_dma_conf_mbus_windows(void __iomem *base, int win, /* try to find matching cs for current dma address */ for (i = 0; i < dram->num_cs; i++) { - const struct mbus_dram_window *cs = dram->cs + i; + const struct mbus_dram_window *cs = &dram->cs[i]; if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { writel(cs->base & 0xffff0000, base + KIRKWOOD_AUDIO_WIN_BASE_REG(win)); -- 2.39.0

1 0

[PATCH AUTOSEL 5.4 07/15] ASoC: kirkwood: Iterate over array indexes instead of using pointer math
by Sasha Levin 27 Feb '23

27 Feb '23

From: Kees Cook <keescook(a)chromium.org> [ Upstream commit b3bcedc0402fcdc5c8624c433562d9d1882749d8 ] Walking the dram->cs array was seen as accesses beyond the first array item by the compiler. Instead, use the array index directly. This allows for run-time bounds checking under CONFIG_UBSAN_BOUNDS as well. Seen with GCC 13 with -fstrict-flex-arrays: ../sound/soc/kirkwood/kirkwood-dma.c: In function 'kirkwood_dma_conf_mbus_windows.constprop': ../sound/soc/kirkwood/kirkwood-dma.c:90:24: warning: array subscript 0 is outside array bounds of 'const struct mbus_dram_window[0]' [-Warray-bounds=] 90 | if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { | ~~^~~~~~ Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> Link: https://lore.kernel.org/r/20230127224128.never.410-kees@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index d2d5c25bf5502..0215d187bdff5 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -86,7 +86,7 @@ kirkwood_dma_conf_mbus_windows(void __iomem *base, int win, /* try to find matching cs for current dma address */ for (i = 0; i < dram->num_cs; i++) { - const struct mbus_dram_window *cs = dram->cs + i; + const struct mbus_dram_window *cs = &dram->cs[i]; if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { writel(cs->base & 0xffff0000, base + KIRKWOOD_AUDIO_WIN_BASE_REG(win)); -- 2.39.0

1 0

[PATCH AUTOSEL 5.10 09/19] ASoC: kirkwood: Iterate over array indexes instead of using pointer math
by Sasha Levin 27 Feb '23

27 Feb '23

From: Kees Cook <keescook(a)chromium.org> [ Upstream commit b3bcedc0402fcdc5c8624c433562d9d1882749d8 ] Walking the dram->cs array was seen as accesses beyond the first array item by the compiler. Instead, use the array index directly. This allows for run-time bounds checking under CONFIG_UBSAN_BOUNDS as well. Seen with GCC 13 with -fstrict-flex-arrays: ../sound/soc/kirkwood/kirkwood-dma.c: In function 'kirkwood_dma_conf_mbus_windows.constprop': ../sound/soc/kirkwood/kirkwood-dma.c:90:24: warning: array subscript 0 is outside array bounds of 'const struct mbus_dram_window[0]' [-Warray-bounds=] 90 | if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { | ~~^~~~~~ Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> Link: https://lore.kernel.org/r/20230127224128.never.410-kees@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index e037826b24517..2d41e6ab2ce4e 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -86,7 +86,7 @@ kirkwood_dma_conf_mbus_windows(void __iomem *base, int win, /* try to find matching cs for current dma address */ for (i = 0; i < dram->num_cs; i++) { - const struct mbus_dram_window *cs = dram->cs + i; + const struct mbus_dram_window *cs = &dram->cs[i]; if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { writel(cs->base & 0xffff0000, base + KIRKWOOD_AUDIO_WIN_BASE_REG(win)); -- 2.39.0

1 0

[PATCH AUTOSEL 5.15 14/25] ASoC: kirkwood: Iterate over array indexes instead of using pointer math
by Sasha Levin 27 Feb '23

27 Feb '23

From: Kees Cook <keescook(a)chromium.org> [ Upstream commit b3bcedc0402fcdc5c8624c433562d9d1882749d8 ] Walking the dram->cs array was seen as accesses beyond the first array item by the compiler. Instead, use the array index directly. This allows for run-time bounds checking under CONFIG_UBSAN_BOUNDS as well. Seen with GCC 13 with -fstrict-flex-arrays: ../sound/soc/kirkwood/kirkwood-dma.c: In function 'kirkwood_dma_conf_mbus_windows.constprop': ../sound/soc/kirkwood/kirkwood-dma.c:90:24: warning: array subscript 0 is outside array bounds of 'const struct mbus_dram_window[0]' [-Warray-bounds=] 90 | if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { | ~~^~~~~~ Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> Link: https://lore.kernel.org/r/20230127224128.never.410-kees@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index 700a18561a940..640cebd2983e2 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -86,7 +86,7 @@ kirkwood_dma_conf_mbus_windows(void __iomem *base, int win, /* try to find matching cs for current dma address */ for (i = 0; i < dram->num_cs; i++) { - const struct mbus_dram_window *cs = dram->cs + i; + const struct mbus_dram_window *cs = &dram->cs[i]; if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { writel(cs->base & 0xffff0000, base + KIRKWOOD_AUDIO_WIN_BASE_REG(win)); -- 2.39.0

1 0

[PATCH AUTOSEL 5.15 13/25] ASoC: soc-compress: Reposition and add pcm_mutex
by Sasha Levin 27 Feb '23

27 Feb '23

From: 강신형 <s47.kang(a)samsung.com> [ Upstream commit aa9ff6a4955fdba02b54fbc4386db876603703b7 ] If panic_on_warn is set and compress stream(DPCM) is started, then kernel panic occurred because card->pcm_mutex isn't held appropriately. In the following functions, warning were issued at this line "snd_soc_dpcm_mutex_assert_held". static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { ... snd_soc_dpcm_mutex_assert_held(fe); ... } void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) { ... snd_soc_dpcm_mutex_assert_held(fe); ... } void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd, int stream, int action) { ... snd_soc_dpcm_mutex_assert_held(rtd); ... } int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, int event) { ... snd_soc_dpcm_mutex_assert_held(fe); ... } These functions are called by soc_compr_set_params_fe, soc_compr_open_fe and soc_compr_free_fe without pcm_mutex locking. And this is call stack. [ 414.527841][ T2179] pc : dpcm_process_paths+0x5a4/0x750 [ 414.527848][ T2179] lr : dpcm_process_paths+0x37c/0x750 [ 414.527945][ T2179] Call trace: [ 414.527949][ T2179] dpcm_process_paths+0x5a4/0x750 [ 414.527955][ T2179] soc_compr_open_fe+0xb0/0x2cc [ 414.527972][ T2179] snd_compr_open+0x180/0x248 [ 414.527981][ T2179] snd_open+0x15c/0x194 [ 414.528003][ T2179] chrdev_open+0x1b0/0x220 [ 414.528023][ T2179] do_dentry_open+0x30c/0x594 [ 414.528045][ T2179] vfs_open+0x34/0x44 [ 414.528053][ T2179] path_openat+0x914/0xb08 [ 414.528062][ T2179] do_filp_open+0xc0/0x170 [ 414.528068][ T2179] do_sys_openat2+0x94/0x18c [ 414.528076][ T2179] __arm64_sys_openat+0x78/0xa4 [ 414.528084][ T2179] invoke_syscall+0x48/0x10c [ 414.528094][ T2179] el0_svc_common+0xbc/0x104 [ 414.528099][ T2179] do_el0_svc+0x34/0xd8 [ 414.528103][ T2179] el0_svc+0x34/0xc4 [ 414.528125][ T2179] el0t_64_sync_handler+0x8c/0xfc [ 414.528133][ T2179] el0t_64_sync+0x1a0/0x1a4 [ 414.528142][ T2179] Kernel panic - not syncing: panic_on_warn set ... So, I reposition and add pcm_mutex to resolve lockdep error. Signed-off-by: Shinhyung Kang <s47.kang(a)samsung.com> Link: https://lore.kernel.org/r/016401d90ac4$7b6848c0$7238da40$@samsung.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-compress.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 2050728063a15..9146bec146c1e 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -116,6 +116,8 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) if (ret < 0) goto be_err; + mutex_lock_nested(&fe->card->pcm_mutex, fe->card->pcm_subclass); + /* calculate valid and active FE <-> BE dpcms */ dpcm_process_paths(fe, stream, &list, 1); fe->dpcm[stream].runtime = fe_substream->runtime; @@ -151,7 +153,6 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) fe->dpcm[stream].state = SND_SOC_DPCM_STATE_OPEN; fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; - mutex_lock_nested(&fe->card->pcm_mutex, fe->card->pcm_subclass); snd_soc_runtime_activate(fe, stream); mutex_unlock(&fe->card->pcm_mutex); @@ -182,7 +183,6 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream) mutex_lock_nested(&fe->card->pcm_mutex, fe->card->pcm_subclass); snd_soc_runtime_deactivate(fe, stream); - mutex_unlock(&fe->card->pcm_mutex); fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; @@ -201,6 +201,8 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream) dpcm_be_disconnect(fe, stream); + mutex_unlock(&fe->card->pcm_mutex); + fe->dpcm[stream].runtime = NULL; snd_soc_link_compr_shutdown(cstream, 0); @@ -376,8 +378,9 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, ret = snd_soc_link_compr_set_params(cstream); if (ret < 0) goto out; - + mutex_lock_nested(&fe->card->pcm_mutex, fe->card->pcm_subclass); dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_START); + mutex_unlock(&fe->card->pcm_mutex); fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE; out: -- 2.39.0

1 0

[PATCH AUTOSEL 6.1 34/58] ASoC: kirkwood: Iterate over array indexes instead of using pointer math
by Sasha Levin 27 Feb '23

27 Feb '23

From: Kees Cook <keescook(a)chromium.org> [ Upstream commit b3bcedc0402fcdc5c8624c433562d9d1882749d8 ] Walking the dram->cs array was seen as accesses beyond the first array item by the compiler. Instead, use the array index directly. This allows for run-time bounds checking under CONFIG_UBSAN_BOUNDS as well. Seen with GCC 13 with -fstrict-flex-arrays: ../sound/soc/kirkwood/kirkwood-dma.c: In function 'kirkwood_dma_conf_mbus_windows.constprop': ../sound/soc/kirkwood/kirkwood-dma.c:90:24: warning: array subscript 0 is outside array bounds of 'const struct mbus_dram_window[0]' [-Warray-bounds=] 90 | if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { | ~~^~~~~~ Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> Link: https://lore.kernel.org/r/20230127224128.never.410-kees@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index 700a18561a940..640cebd2983e2 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -86,7 +86,7 @@ kirkwood_dma_conf_mbus_windows(void __iomem *base, int win, /* try to find matching cs for current dma address */ for (i = 0; i < dram->num_cs; i++) { - const struct mbus_dram_window *cs = dram->cs + i; + const struct mbus_dram_window *cs = &dram->cs[i]; if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { writel(cs->base & 0xffff0000, base + KIRKWOOD_AUDIO_WIN_BASE_REG(win)); -- 2.39.0

1 0

[PATCH AUTOSEL 6.1 33/58] ASoC: soc-compress: Reposition and add pcm_mutex
by Sasha Levin 27 Feb '23

27 Feb '23

From: 강신형 <s47.kang(a)samsung.com> [ Upstream commit aa9ff6a4955fdba02b54fbc4386db876603703b7 ] If panic_on_warn is set and compress stream(DPCM) is started, then kernel panic occurred because card->pcm_mutex isn't held appropriately. In the following functions, warning were issued at this line "snd_soc_dpcm_mutex_assert_held". static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { ... snd_soc_dpcm_mutex_assert_held(fe); ... } void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) { ... snd_soc_dpcm_mutex_assert_held(fe); ... } void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd, int stream, int action) { ... snd_soc_dpcm_mutex_assert_held(rtd); ... } int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, int event) { ... snd_soc_dpcm_mutex_assert_held(fe); ... } These functions are called by soc_compr_set_params_fe, soc_compr_open_fe and soc_compr_free_fe without pcm_mutex locking. And this is call stack. [ 414.527841][ T2179] pc : dpcm_process_paths+0x5a4/0x750 [ 414.527848][ T2179] lr : dpcm_process_paths+0x37c/0x750 [ 414.527945][ T2179] Call trace: [ 414.527949][ T2179] dpcm_process_paths+0x5a4/0x750 [ 414.527955][ T2179] soc_compr_open_fe+0xb0/0x2cc [ 414.527972][ T2179] snd_compr_open+0x180/0x248 [ 414.527981][ T2179] snd_open+0x15c/0x194 [ 414.528003][ T2179] chrdev_open+0x1b0/0x220 [ 414.528023][ T2179] do_dentry_open+0x30c/0x594 [ 414.528045][ T2179] vfs_open+0x34/0x44 [ 414.528053][ T2179] path_openat+0x914/0xb08 [ 414.528062][ T2179] do_filp_open+0xc0/0x170 [ 414.528068][ T2179] do_sys_openat2+0x94/0x18c [ 414.528076][ T2179] __arm64_sys_openat+0x78/0xa4 [ 414.528084][ T2179] invoke_syscall+0x48/0x10c [ 414.528094][ T2179] el0_svc_common+0xbc/0x104 [ 414.528099][ T2179] do_el0_svc+0x34/0xd8 [ 414.528103][ T2179] el0_svc+0x34/0xc4 [ 414.528125][ T2179] el0t_64_sync_handler+0x8c/0xfc [ 414.528133][ T2179] el0t_64_sync+0x1a0/0x1a4 [ 414.528142][ T2179] Kernel panic - not syncing: panic_on_warn set ... So, I reposition and add pcm_mutex to resolve lockdep error. Signed-off-by: Shinhyung Kang <s47.kang(a)samsung.com> Link: https://lore.kernel.org/r/016401d90ac4$7b6848c0$7238da40$@samsung.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-compress.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 870f13e1d389c..7bce5088b4554 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -149,6 +149,8 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) if (ret < 0) goto be_err; + mutex_lock_nested(&fe->card->pcm_mutex, fe->card->pcm_subclass); + /* calculate valid and active FE <-> BE dpcms */ dpcm_process_paths(fe, stream, &list, 1); fe->dpcm[stream].runtime = fe_substream->runtime; @@ -184,7 +186,6 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) fe->dpcm[stream].state = SND_SOC_DPCM_STATE_OPEN; fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; - mutex_lock_nested(&fe->card->pcm_mutex, fe->card->pcm_subclass); snd_soc_runtime_activate(fe, stream); mutex_unlock(&fe->card->pcm_mutex); @@ -215,7 +216,6 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream) mutex_lock_nested(&fe->card->pcm_mutex, fe->card->pcm_subclass); snd_soc_runtime_deactivate(fe, stream); - mutex_unlock(&fe->card->pcm_mutex); fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; @@ -234,6 +234,8 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream) dpcm_be_disconnect(fe, stream); + mutex_unlock(&fe->card->pcm_mutex); + fe->dpcm[stream].runtime = NULL; snd_soc_link_compr_shutdown(cstream, 0); @@ -409,8 +411,9 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, ret = snd_soc_link_compr_set_params(cstream); if (ret < 0) goto out; - + mutex_lock_nested(&fe->card->pcm_mutex, fe->card->pcm_subclass); dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_START); + mutex_unlock(&fe->card->pcm_mutex); fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE; out: -- 2.39.0

1 0

[PATCH AUTOSEL 6.2 35/60] ASoC: kirkwood: Iterate over array indexes instead of using pointer math
by Sasha Levin 27 Feb '23

27 Feb '23

From: Kees Cook <keescook(a)chromium.org> [ Upstream commit b3bcedc0402fcdc5c8624c433562d9d1882749d8 ] Walking the dram->cs array was seen as accesses beyond the first array item by the compiler. Instead, use the array index directly. This allows for run-time bounds checking under CONFIG_UBSAN_BOUNDS as well. Seen with GCC 13 with -fstrict-flex-arrays: ../sound/soc/kirkwood/kirkwood-dma.c: In function 'kirkwood_dma_conf_mbus_windows.constprop': ../sound/soc/kirkwood/kirkwood-dma.c:90:24: warning: array subscript 0 is outside array bounds of 'const struct mbus_dram_window[0]' [-Warray-bounds=] 90 | if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { | ~~^~~~~~ Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> Link: https://lore.kernel.org/r/20230127224128.never.410-kees@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index 700a18561a940..640cebd2983e2 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -86,7 +86,7 @@ kirkwood_dma_conf_mbus_windows(void __iomem *base, int win, /* try to find matching cs for current dma address */ for (i = 0; i < dram->num_cs; i++) { - const struct mbus_dram_window *cs = dram->cs + i; + const struct mbus_dram_window *cs = &dram->cs[i]; if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) { writel(cs->base & 0xffff0000, base + KIRKWOOD_AUDIO_WIN_BASE_REG(win)); -- 2.39.0

1 0

[PATCH AUTOSEL 6.2 34/60] ASoC: soc-compress: Reposition and add pcm_mutex
by Sasha Levin 27 Feb '23

27 Feb '23

From: 강신형 <s47.kang(a)samsung.com> [ Upstream commit aa9ff6a4955fdba02b54fbc4386db876603703b7 ] If panic_on_warn is set and compress stream(DPCM) is started, then kernel panic occurred because card->pcm_mutex isn't held appropriately. In the following functions, warning were issued at this line "snd_soc_dpcm_mutex_assert_held". static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { ... snd_soc_dpcm_mutex_assert_held(fe); ... } void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) { ... snd_soc_dpcm_mutex_assert_held(fe); ... } void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd, int stream, int action) { ... snd_soc_dpcm_mutex_assert_held(rtd); ... } int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, int event) { ... snd_soc_dpcm_mutex_assert_held(fe); ... } These functions are called by soc_compr_set_params_fe, soc_compr_open_fe and soc_compr_free_fe without pcm_mutex locking. And this is call stack. [ 414.527841][ T2179] pc : dpcm_process_paths+0x5a4/0x750 [ 414.527848][ T2179] lr : dpcm_process_paths+0x37c/0x750 [ 414.527945][ T2179] Call trace: [ 414.527949][ T2179] dpcm_process_paths+0x5a4/0x750 [ 414.527955][ T2179] soc_compr_open_fe+0xb0/0x2cc [ 414.527972][ T2179] snd_compr_open+0x180/0x248 [ 414.527981][ T2179] snd_open+0x15c/0x194 [ 414.528003][ T2179] chrdev_open+0x1b0/0x220 [ 414.528023][ T2179] do_dentry_open+0x30c/0x594 [ 414.528045][ T2179] vfs_open+0x34/0x44 [ 414.528053][ T2179] path_openat+0x914/0xb08 [ 414.528062][ T2179] do_filp_open+0xc0/0x170 [ 414.528068][ T2179] do_sys_openat2+0x94/0x18c [ 414.528076][ T2179] __arm64_sys_openat+0x78/0xa4 [ 414.528084][ T2179] invoke_syscall+0x48/0x10c [ 414.528094][ T2179] el0_svc_common+0xbc/0x104 [ 414.528099][ T2179] do_el0_svc+0x34/0xd8 [ 414.528103][ T2179] el0_svc+0x34/0xc4 [ 414.528125][ T2179] el0t_64_sync_handler+0x8c/0xfc [ 414.528133][ T2179] el0t_64_sync+0x1a0/0x1a4 [ 414.528142][ T2179] Kernel panic - not syncing: panic_on_warn set ... So, I reposition and add pcm_mutex to resolve lockdep error. Signed-off-by: Shinhyung Kang <s47.kang(a)samsung.com> Link: https://lore.kernel.org/r/016401d90ac4$7b6848c0$7238da40$@samsung.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-compress.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 870f13e1d389c..7bce5088b4554 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -149,6 +149,8 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) if (ret < 0) goto be_err; + mutex_lock_nested(&fe->card->pcm_mutex, fe->card->pcm_subclass); + /* calculate valid and active FE <-> BE dpcms */ dpcm_process_paths(fe, stream, &list, 1); fe->dpcm[stream].runtime = fe_substream->runtime; @@ -184,7 +186,6 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) fe->dpcm[stream].state = SND_SOC_DPCM_STATE_OPEN; fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; - mutex_lock_nested(&fe->card->pcm_mutex, fe->card->pcm_subclass); snd_soc_runtime_activate(fe, stream); mutex_unlock(&fe->card->pcm_mutex); @@ -215,7 +216,6 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream) mutex_lock_nested(&fe->card->pcm_mutex, fe->card->pcm_subclass); snd_soc_runtime_deactivate(fe, stream); - mutex_unlock(&fe->card->pcm_mutex); fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; @@ -234,6 +234,8 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream) dpcm_be_disconnect(fe, stream); + mutex_unlock(&fe->card->pcm_mutex); + fe->dpcm[stream].runtime = NULL; snd_soc_link_compr_shutdown(cstream, 0); @@ -409,8 +411,9 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, ret = snd_soc_link_compr_set_params(cstream); if (ret < 0) goto out; - + mutex_lock_nested(&fe->card->pcm_mutex, fe->card->pcm_subclass); dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_START); + mutex_unlock(&fe->card->pcm_mutex); fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE; out: -- 2.39.0

1 0

Alsamixer not detecting headphones
by GitHub issues - opened 27 Feb '23

27 Feb '23

alsa-project/alsa-ucm-conf issue #283 was opened from lgmventura: I'm using the motherboard ASUS ROG Strix B760-f Gaming Wifi, which has 7.1 audio channels on the back side and connectors for an audio panel at the front of the case - a microphone and a headphones jack. When I plug my headphones into the front channels connector (at the back of the PC case) there is normal sound. When I plug it into the headphones jack (at the front of the case), it gets automatically detected by `pavucontrol`, which switches to the headphones as output, as it should be: ![image](https://user-images.githubusercontent.com/16710238/221439991-903fd90b-43b2-41a0-af7f-1ed02085238e.png) However, there is no sound. When I open `alsamixer` and choose the respective soundcard, which is weirdly called "USB audio" there, I see no volume control for the headphones, as there should be. All it shows is this: ![image](https://user-images.githubusercontent.com/16710238/221440193-381eadfb-fed9-48ad-807b-05c736b65b24.png) The only way I got to get audio output from the headphones jack is by running `aplay` and choosing the hardware directly from there `aplay -D plughw:1,1 /usr/share/sounds/alsa/Front_Center.wav` or on Audacity by selecting "USB Audio 1: (hw:1,1)" as playback device. I also tried to use JACK Audio Connection Kit, but found no configuration there that routes the audio to the headphones jack, even trying all 4 possible analogue outputs. System: - Motherboard ASUS ROG Strix B760-f Gaming Wifi - Intel core i7-13700K - Ubuntu 22.04 (freshly installed) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/283 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Alsamixer not detecting headphones
by GitHub issues - edited 27 Feb '23

27 Feb '23

alsa-project/alsa-ucm-conf issue #283 was edited from lgmventura: I'm using the motherboard ASUS ROG Strix B760-f Gaming Wifi, which has 7.1 audio channels on the back side and connectors for an audio panel at the front of the case - a microphone and a headphones jack. When I plug my headphones into the front channels connector (at the back of the PC case) there is normal sound. When I plug it into the headphones jack (at the front of the case), it gets automatically detected by `pavucontrol`, which switches to the headphones as output, as it should be: ![image](https://user-images.githubusercontent.com/16710238/221439991-903fd90b-43b2-41a0-af7f-1ed02085238e.png) However, there is no sound. When I open `alsamixer` and choose the respective soundcard, which is weirdly called "USB audio" there, I see no volume control for the headphones, as there should be. All it shows is this: ![image](https://user-images.githubusercontent.com/16710238/221440193-381eadfb-fed9-48ad-807b-05c736b65b24.png) The only way I got to get audio output from the headphones jack is by running `aplay` and choosing the hardware directly from there `aplay -D plughw:1,1 /usr/share/sounds/alsa/Front_Center.wav` or on Audacity by selecting "USB Audio 1: (hw:1,1)" as playback device. I also tried to use JACK Audio Connection Kit, but found no configuration there that routes the audio to the headphones jack, even trying all 4 possible analogue outputs. System: - Motherboard ASUS ROG Strix B760-f Gaming Wifi - Intel core i7-13700K - Ubuntu 22.04 (freshly installed) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/283 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Alsamixer not detecting headphones
by GitHub issues - opened 27 Feb '23

27 Feb '23

alsa-project/alsa-lib issue #304 was opened from lgmventura: I'm using the motherboard ASUS ROG Strix B760-f Gaming Wifi, which has 7.1 audio channels on the back side and connectors for an audio panel at the front of the case - a microphone and a headphones jack. When I plug my headphones into the front channels connector (at the back of the PC case) there is normal sound. When I plug it into the headphones jack (at the front of the case), it gets automatically detected by `pavucontrol`, which switches to the headphones as output, as it should be: ![image](https://user-images.githubusercontent.com/16710238/221439991-903fd90b-43b2-41a0-af7f-1ed02085238e.png) However, there is no sound. When I open `alsamixer` and choose the respective soundcard, which is weirdly called "USB audio" there, I see no volume control for the headphones, as there should be. All it shows is this: ![image](https://user-images.githubusercontent.com/16710238/221440193-381eadfb-fed9-48ad-807b-05c736b65b24.png) The only way I got to get audio output from the headphones jack is by running `aplay` and choosing the hardware directly from there `aplay -D plughw:1,1 /usr/share/sounds/alsa/Front_Center.wav` or on Audacity by selecting "USB Audio 1: (hw:1,1)" as playback device. I also tried to use JACK Audio Connection Kit, but found no configuration there that routes the audio to the headphones jack, even trying all 4 possible analogue outputs. System: - Motherboard ASUS ROG Strix B760-f Gaming Wifi - Intel core i7-13700K - Ubuntu 22.04 (freshly installed) Issue URL : https://github.com/alsa-project/alsa-lib/issues/304 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

infinite loop in src/pcm/pcm_params.c function snd_pcm_sw_params_default()
by bumpelo＠gmail.com 25 Feb '23

25 Feb '23

I found this infinite loop while playing with the test/latency.c function. When I specify too small of a minimum latency (in my case -m 256 or smaller) it winds up: at line line 2337: params->boundary = pcm->buffer_size; while (params->boundary * 2 <= LONG_MAX - pcm->buffer_size) params->boundary *= 2; return 0; but when pcm->buffer_size is zero this is an infinite loop. It does this *after* setting the scheduling to round-robin which makes the infinite loop so much more painful. The return value from this function is ignored where it's called so I'm not sure what the best fix is. I'm guessing it's a rare case since it's not been previously reported but it deserves fixing anyway.

2 1

xone mixer
by GitHub issues - opened 25 Feb '23

25 Feb '23

alsa-project/alsa-lib issue #303 was opened from molli1312: Having a few problems with the Xone32c though not sure if this is an ALSA issue or general USB stack problem. It's a USB soundcard that is supposed to have 4 inputs and 4 outputs. It seems to have 2 USB configurations (bNumConfigurations : 0x2). When plugged in, bConfigurationValue is 0x1 and only 2 inputs and 2 outputs show up. Change this value to 0x2 and it shows up (as it should) as 4 inputs and 4 outputs. I can change the bConfigurationValue with this udev rule, or with a `echo 2 /sys/bus/usb/devices/1-1.4/bConfigurationValue`: This has all been working OK on Ubuntu 20.04 running kernel 5.4.0-33-generic. However, I just updated to 5.4.0-37-generic and now there is only a single USB configuration (bNumConfigurations : 0x1) and bConfigurationValue can't be changed to 0x2: Issue URL : https://github.com/alsa-project/alsa-lib/issues/303 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

OpenBSD (and maybe other *BSDs) support (2)
by GitHub pull_request - opened 25 Feb '23

25 Feb '23

alsa-project/alsa-utils pull request #193 was opened from jg1uaa: #186 addendum. forget to fix alsactl/lock.c, this is the fix. Request URL : https://github.com/alsa-project/alsa-utils/pull/193 Patch URL : https://github.com/alsa-project/alsa-utils/pull/193.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

hda-verb improvements for Asus Zenbook Pro Duo UX582LR
by Marcel Brouillet 25 Feb '23

25 Feb '23

Hello, I have a Asus Zenbook Pro Duo UX582LR that does not have sound out-of-the box. But with either this : hda-verb /dev/snd/hwC0D0 0x20 0x500 0xf hda-verb /dev/snd/hwC0D0 0x20 0x400 0x7774 hda-verb /dev/snd/hwC0D0 0x20 0x500 0x45 hda-verb /dev/snd/hwC0D0 0x20 0x400 0x5289 Or this : hda-verb /dev/snd/hwC0D0 0x20 0x500 0x1b hda-verb /dev/snd/hwC0D0 0x20 0x477 0x4a4b hda-verb /dev/snd/hwC0D0 0x20 0x500 0xf hda-verb /dev/snd/hwC0D0 0x20 0x477 0x74 Then 1. I do get output from the laptop speakers ✅ 2. I do get output through a headset when pluged in (and no sound through speakers) ✅ 3. I do get input from laptop mics, *but far too high gain. *Setting the mic level to 35% in Pulse Audio is roughly good 4. Input *remains from laptop mics even when I plug in headsets* (TRRS jack) I've been OK with this for 18 months, but decided to chip in the time to solve 3 and 4 and make it available to the community. I looked at https://github.com/alsa-project/alsa-tools/tree/master/hda-verb and understand the hda-verb syntax. But *where do I find the verb arguments (strings or the above cryptic 0x477 or 0x400) ?* Do I need to read this 220 pages document ? https://www.intel.com/content/www/us/en/standards/high-definition-audio-spe… *If anyone is willing to direct me to the few verbs to try, and please tell me if there is any risk of killing my sound interface with wrong parameters, a big thank you to you.* Regards, Marcel. Hardware : Asus Zenbook Pro Duo UX582LR Software : Ubuntu Studio 22.04.2 LTS jammy. Fresh installation 10 days ago, fully updated upgraded. alsa-base/jammy,jammy,now 1.0.25+dfsg-0ubuntu7 all [installed,automatic] alsa-tools/jammy,now 1.2.5-2 amd64 [installed,automatic] alsa-tools-gui/jammy,now 1.2.5-2 amd64 [installed,automatic] alsa-topology-conf/jammy,jammy,now 1.2.5.1-2 all [installed,automatic] alsa-ucm-conf/jammy-updates,jammy-updates,now 1.2.6.3-1ubuntu1.4 all [installed,automatic] alsa-utils/jammy,now 1.2.6-1ubuntu1 amd64 [installed,automatic]

1 0

[PATCH 1/3] ASoC: apple: mca: Fix final status read on SERDES reset
by Martin Povišer 24 Feb '23

24 Feb '23

From within the early trigger we are doing a reset of the SERDES unit, but the final status read is on a bad address. Add the missing SERDES unit offset in calculation of the address. Fixes: 3df5d0d97289 ("ASoC: apple: mca: Start new platform driver") Signed-off-by: Martin Povišer <povik+lin(a)cutebit.org> --- sound/soc/apple/mca.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/apple/mca.c b/sound/soc/apple/mca.c index 24381c42eb54..9cceeb259952 100644 --- a/sound/soc/apple/mca.c +++ b/sound/soc/apple/mca.c @@ -210,7 +210,7 @@ static void mca_fe_early_trigger(struct snd_pcm_substream *substream, int cmd, SERDES_CONF_SOME_RST); readl_relaxed(cl->base + serdes_conf); mca_modify(cl, serdes_conf, SERDES_STATUS_RST, 0); - WARN_ON(readl_relaxed(cl->base + REG_SERDES_STATUS) & + WARN_ON(readl_relaxed(cl->base + serdes_unit + REG_SERDES_STATUS) & SERDES_STATUS_RST); break; default: -- 2.33.0

2 3

[PATCH] ASoC: adau7118: don't disable regulators on device unbind
by Nuno Sá 24 Feb '23

24 Feb '23

The regulators are supposed to be controlled through the set_bias_level() component callback. Moreover, the regulators are not enabled during probe and so, this would lead to a regulator unbalanced use count. Fixes: ca514c0f12b02 ("ASOC: Add ADAU7118 8 Channel PDM-to-I2S/TDM Converter driver") Signed-off-by: Nuno Sá <nuno.sa(a)analog.com> --- Note this was only checked by code inspection but unless I'm missing something, it looks clear this could happen by just insmod + rmmod the module. sound/soc/codecs/adau7118.c | 19 +------------------ 1 file changed, 1 insertion(+), 18 deletions(-) diff --git a/sound/soc/codecs/adau7118.c b/sound/soc/codecs/adau7118.c index bbb097249887..a663d37e5776 100644 --- a/sound/soc/codecs/adau7118.c +++ b/sound/soc/codecs/adau7118.c @@ -444,22 +444,6 @@ static const struct snd_soc_component_driver adau7118_component_driver = { .endianness = 1, }; -static void adau7118_regulator_disable(void *data) -{ - struct adau7118_data *st = data; - int ret; - /* - * If we fail to disable DVDD, don't bother in trying IOVDD. We - * actually don't want to be left in the situation where DVDD - * is enabled and IOVDD is disabled. - */ - ret = regulator_disable(st->dvdd); - if (ret) - return; - - regulator_disable(st->iovdd); -} - static int adau7118_regulator_setup(struct adau7118_data *st) { st->iovdd = devm_regulator_get(st->dev, "iovdd"); @@ -481,8 +465,7 @@ static int adau7118_regulator_setup(struct adau7118_data *st) regcache_cache_only(st->map, true); } - return devm_add_action_or_reset(st->dev, adau7118_regulator_disable, - st); + return 0; } static int adau7118_parset_dt(const struct adau7118_data *st) -- 2.39.2

2 1

[syzbot] [alsa?] kernel BUG in __vunmap
by syzbot 23 Feb '23

23 Feb '23

Hello, syzbot found the following issue on: HEAD commit: a5541c0811a0 Merge branch 'for-next/core' into for-kernelci git tree: git://git.kernel.org/pub/scm/linux/kernel/git/arm64/linux.git for-kernelci console output: https://syzkaller.appspot.com/x/log.txt?x=12b4018b880000 kernel config: https://syzkaller.appspot.com/x/.config?x=cbd4e584773e9397 dashboard link: https://syzkaller.appspot.com/bug?extid=656c0cde55a9d107a76f compiler: Debian clang version 13.0.1-++20220126092033+75e33f71c2da-1~exp1~20220126212112.63, GNU ld (GNU Binutils for Debian) 2.35.2 userspace arch: arm64 Unfortunately, I don't have any reproducer for this issue yet. Downloadable assets: disk image: https://storage.googleapis.com/syzbot-assets/4b7702208fb9/disk-a5541c08.raw… vmlinux: https://storage.googleapis.com/syzbot-assets/9ec0153ec051/vmlinux-a5541c08.… kernel image: https://storage.googleapis.com/syzbot-assets/6f8725ad290a/Image-a5541c08.gz… IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+656c0cde55a9d107a76f(a)syzkaller.appspotmail.com ------------[ cut here ]------------ kernel BUG at mm/vmalloc.c:2707! Internal error: Oops - BUG: 00000000f2000800 [#1] PREEMPT SMP Modules linked in: CPU: 1 PID: 9210 Comm: syz-executor.0 Not tainted 6.1.0-rc8-syzkaller-33330-ga5541c0811a0 #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 10/26/2022 pstate: 80400005 (Nzcv daif +PAN -UAO -TCO -DIT -SSBS BTYPE=--) pc : __vunmap+0x488/0x4cc mm/vmalloc.c:2707 lr : __vunmap+0x488/0x4cc mm/vmalloc.c:2707 sp : ffff800013d73ba0 x29: ffff800013d73bb0 x28: 00000000000c0002 x27: 0000000000002000 x26: ffff0000c5a44d68 x25: ffff0000c6a14080 x24: ffff00012b5ed000 x23: 0000000000000001 x22: 0000000000000200 x21: 0000000000000000 x20: ffff80000cbdd3b4 x19: ffff0001268cfa00 x18: 0000000000000000 x17: 0000000000000000 x16: ffff80000dbe6158 x15: ffff000114e01a40 x14: 0000000000000000 x13: 0000000000000000 x12: 0000000000000000 x11: ff808000084b647c x10: 0000000000000000 x9 : ffff8000084b647c x8 : ffff000114e01a40 x7 : ffff8000084bf4d4 x6 : 0000000000000000 x5 : 0000000000000080 x4 : 0000000000000000 x3 : 0000000000000000 x2 : 0000000000000000 x1 : ffff80000cbaae40 x0 : 0000000000000000 Call trace: __vunmap+0x488/0x4cc mm/vmalloc.c:2707 __vfree mm/vmalloc.c:2761 [inline] vfree+0xb0/0xfc mm/vmalloc.c:2792 snd_dma_vmalloc_free+0x20/0x30 sound/core/memalloc.c:356 snd_dma_free_pages+0x84/0xc0 sound/core/memalloc.c:127 do_free_pages sound/core/pcm_memory.c:65 [inline] snd_pcm_lib_free_pages+0xa0/0x12c sound/core/pcm_memory.c:471 do_hw_free sound/core/pcm_native.c:876 [inline] snd_pcm_release_substream+0x15c/0x1dc sound/core/pcm_native.c:2705 snd_pcm_oss_release_file sound/core/oss/pcm_oss.c:2413 [inline] snd_pcm_oss_release+0x98/0x130 sound/core/oss/pcm_oss.c:2592 __fput+0x198/0x3e4 fs/file_table.c:320 ____fput+0x20/0x30 fs/file_table.c:348 task_work_run+0x100/0x148 kernel/task_work.c:179 resume_user_mode_work include/linux/resume_user_mode.h:49 [inline] do_notify_resume+0x174/0x1f0 arch/arm64/kernel/signal.c:1132 prepare_exit_to_user_mode arch/arm64/kernel/entry-common.c:137 [inline] exit_to_user_mode arch/arm64/kernel/entry-common.c:142 [inline] el0_svc+0x9c/0x150 arch/arm64/kernel/entry-common.c:638 el0t_64_sync_handler+0x84/0xf0 arch/arm64/kernel/entry-common.c:655 el0t_64_sync+0x190/0x194 arch/arm64/kernel/entry.S:584 Code: 17ffff1f 97f7c16d 17ffff18 97f7c16b (d4210000) ---[ end trace 0000000000000000 ]--- --- This report is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this issue. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot.

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Re: Firewire Audio Interface
by Takashi Sakamoto 23 Feb '23

23 Feb '23

Hi, (C.C.ed to alsa-devel) On Wed, Feb 22, 2023 at 11:51:23AM -0500, mj@mj wrote: > Hey man thanks so much for the reply! > > Why TC put the phantom switch in the software instead of on the > hardware unit I can’t figure out! > > I think I’m just going to use it a 2-in 2-out interface without > the phantom. > > A friend of mine looked at the circuit board and told me it > would be too difficult to put a switch for the phantom power in it. > > Thanks again for the reply! If it is Desktop Konnekt 6, node offset 0x'ffff'e0a0'103c has the 32 bit value to express the state of phantom powering. If it is 0, disabled, else enabled. Quadlet write asynchronous transaction can change the state, but I have never used MacOs to operate any node in IEEE 1394 bus. I know some developer and user for the craft driver; e.g. * Re: [alsa-devel] Tascam Coding Inquiry * https://lore.kernel.org/alsa-devel/20200108095430.GA16741@workstation/ I think it your help to contact to them. Especially Paul Neyrinck[1] seems to complete his work for the craft driver of Digidesign Digi 002/003 family in MacOS. They are different hardware from yours, while he will help you to execute the simple read/write transaction to a certain offset of it. [1] https://neyrinck.com/ Regards Takashi Sakamoto

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[PATCH v6 0/4] Add CS35L41 shared boost feature
by Lucas Tanure 22 Feb '23

22 Feb '23

Valve's Steam Deck uses CS35L41 in shared boost mode, where both speakers share the boost circuit. Add this support in the shared lib, but for now, shared boost is not supported in HDA systems as would require BIOS changes. Based on David Rhodes shared boost patches. Also, fix boost config overwriting in IRQ found in the review and do a small refactor of the code. Changes from V5: - Improved documentation from Cirrrus - All amps use MDSCYN for boost source - Active amp has TX and RX enabled Changes from V4: - Fix Document subject Changes from V3: - Fix wrong code sent - Fix ISO C90 mixed declarations and code Changes from V2: - Drop External boost without VSPK Documentation - Move Shared boost to use values 2 and 3 - Revert back to reg_sequence but reading the value first and only update the necessary bits - Fix bug found by Intel kernel Test Robot Changes from V1: - Fix Documentation patch subject - New patch for External boost without VSPK Documentation - New patch to fix boost IRQ overwriting issue - New patch to refactor IRQ release error code - reinit_completion on pcm_startup - fix DRE switch overwriting - return IRQ_HANDLED in PLL_LOCK case Lucas Tanure (4): ASoC: cs35l41: Only disable internal boost ASoC: cs35l41: Refactor error release code ALSA: cs35l41: Add shared boost feature ASoC: dt-bindings: cirrus,cs35l41: Document CS35l41 shared boost .../bindings/sound/cirrus,cs35l41.yaml | 10 +- include/sound/cs35l41.h | 13 +- sound/pci/hda/cs35l41_hda.c | 6 +- sound/soc/codecs/cs35l41-lib.c | 73 +++++++++- sound/soc/codecs/cs35l41.c | 125 +++++++++--------- sound/soc/codecs/cs35l41.h | 1 + 6 files changed, 157 insertions(+), 71 deletions(-) -- 2.39.2

2 5

[PATCH] ucm2: Rockchip: rk817: Add ALSA UCM support
by Chris Morgan 22 Feb '23

22 Feb '23

From: Chris Morgan <macromorgan(a)hotmail.com> Add ALSA-UCM support for the Rockchip RK817 audio codec. This codec is typically configured either with an internal or external amplifier as reflected by the longname. This configuration has been tested on the Anbernic RG353P (rk817_ext) and the Odroid Go Advance (rk817_int). Signed-off-by: Chris Morgan <macromorgan(a)hotmail.com> --- ucm2/Rockchip/rk817-sound/HiFi.conf | 66 ++++++++++++++++++++++ ucm2/Rockchip/rk817-sound/rk817-sound.conf | 7 +++ ucm2/conf.d/simple-card/rk817_ext.conf | 1 + ucm2/conf.d/simple-card/rk817_int.conf | 1 + 4 files changed, 75 insertions(+) create mode 100644 ucm2/Rockchip/rk817-sound/HiFi.conf create mode 100644 ucm2/Rockchip/rk817-sound/rk817-sound.conf create mode 120000 ucm2/conf.d/simple-card/rk817_ext.conf create mode 120000 ucm2/conf.d/simple-card/rk817_int.conf diff --git a/ucm2/Rockchip/rk817-sound/HiFi.conf b/ucm2/Rockchip/rk817-sound/HiFi.conf new file mode 100644 index 0000000..e285db7 --- /dev/null +++ b/ucm2/Rockchip/rk817-sound/HiFi.conf @@ -0,0 +1,66 @@ +If.1 { + Condition { + Type ControlExists + Control "name='Internal Speakers Switch'" + } + + True { + SectionDevice."Speaker".EnableSequence [ + cset "name='Playback Mux' HP" + cset "name='Internal Speakers Switch' on" + ] + + SectionDevice."Speaker".DisableSequence [ + cset "name='Internal Speakers Switch' off" + ] + } +} + +SectionDevice."Speaker" { + Comment "Internal Speaker" + + EnableSequence [ + cset "name='Playback Mux' SPK" + ] + + Value { + PlaybackMasterElem "Master Playback Volume" + PlaybackPriority 100 + PlaybackPCM "hw:${CardId}" + } + + ConflictingDevice [ + "Headphones" + ] +} + +SectionDevice."Mic" { + Comment "Microphone" + + Value { + CapturePriority 100 + CapturePCM "hw:${CardId}" + CaptureMixerElem "Mic Capture Gain" + CaptureMasterElem "Master Capture Volume" + } +} + +SectionDevice."Headphones" { + Comment "Headphones" + + EnableSequence [ + cset "name='Playback Mux' HP" + ] + + Value { + PlaybackMasterElem "Master Playback Volume" + PlaybackPriority 200 + PlaybackPCM "hw:${CardId}" + JackControl "Headphones Jack" + JackHWMute "Speaker" + } + + ConflictingDevice [ + "Speaker" + ] +} diff --git a/ucm2/Rockchip/rk817-sound/rk817-sound.conf b/ucm2/Rockchip/rk817-sound/rk817-sound.conf new file mode 100644 index 0000000..6e326af --- /dev/null +++ b/ucm2/Rockchip/rk817-sound/rk817-sound.conf @@ -0,0 +1,7 @@ +Syntax 4 + +SectionUseCase."HiFi" { + File "/Rockchip/rk817-sound/HiFi.conf" + Comment "Play HiFi quality music" +} + diff --git a/ucm2/conf.d/simple-card/rk817_ext.conf b/ucm2/conf.d/simple-card/rk817_ext.conf new file mode 120000 index 0000000..6fa8db7 --- /dev/null +++ b/ucm2/conf.d/simple-card/rk817_ext.conf @@ -0,0 +1 @@ +../../Rockchip/rk817-sound/rk817-sound.conf \ No newline at end of file diff --git a/ucm2/conf.d/simple-card/rk817_int.conf b/ucm2/conf.d/simple-card/rk817_int.conf new file mode 120000 index 0000000..6fa8db7 --- /dev/null +++ b/ucm2/conf.d/simple-card/rk817_int.conf @@ -0,0 +1 @@ +../../Rockchip/rk817-sound/rk817-sound.conf \ No newline at end of file -- 2.34.1

2 1

[PATCH 00/10] ASoC: Initial support for Cirrus Logic CS35L56
by Richard Fitzgerald 22 Feb '23

22 Feb '23

The CS35L56 is a high-performance boosted mono audio amplifier. Supported control interfaces are I2C, SPI or SoundWire. Supported audio interfaces are I2S/TDM or SoundWire. The CS35L56 has a self-booting firmware in ROM, with the ability to patch the firmware and/or apply tunings. Patches #1 to #7 add support to cs_dsp and wm_adsp for self-booting firmware and the ability to apply a .bin file without having to apply a .wmfw. Patch #8 adds the ASoC driver. Patches #9 and #10 add support for this to the Intel sof_sdw machine driver. Richard Fitzgerald (3): ASoC: cs35l56: Add driver for Cirrus Logic CS35L56 ASoC: Intel: sof_sdw: Add support for Cirrus Logic CS35L56 ASoC: Intel: soc-acpi: Add CS35L56 Soundwire to TGL Simon Trimmer (7): firmware: cs_dsp: Introduce no_core_startstop for self-booting DSPs ASoC: wm_adsp: Use no_core_startstop to prevent creating preload control firmware: cs_dsp: Support DSPs that don't require firmware download ASoC: wm_adsp: Support DSPs that don't require firmware download ASoC: wm_adsp: Expose the DSP boot work actions as wm_adsp_power_up() ASoC: wm_adsp: Add support for loading bin files without wmfw ASoC: wm_adsp: Simplify the logging of requested firmware files MAINTAINERS | 2 + drivers/firmware/cirrus/cs_dsp.c | 17 +- include/linux/firmware/cirrus/cs_dsp.h | 1 + include/sound/cs35l56.h | 266 +++ sound/soc/codecs/Kconfig | 34 + sound/soc/codecs/Makefile | 10 + sound/soc/codecs/cs35l56-i2c.c | 83 + sound/soc/codecs/cs35l56-sdw.c | 525 ++++++ sound/soc/codecs/cs35l56-shared.c | 387 +++++ sound/soc/codecs/cs35l56-spi.c | 81 + sound/soc/codecs/cs35l56.c | 1451 +++++++++++++++++ sound/soc/codecs/cs35l56.h | 77 + sound/soc/codecs/wm_adsp.c | 63 +- sound/soc/codecs/wm_adsp.h | 3 + sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/Makefile | 1 + sound/soc/intel/boards/sof_sdw.c | 7 + sound/soc/intel/boards/sof_sdw_common.h | 7 + sound/soc/intel/boards/sof_sdw_cs35l56.c | 115 ++ .../intel/common/soc-acpi-intel-tgl-match.c | 57 + 20 files changed, 3172 insertions(+), 16 deletions(-) create mode 100644 include/sound/cs35l56.h create mode 100644 sound/soc/codecs/cs35l56-i2c.c create mode 100644 sound/soc/codecs/cs35l56-sdw.c create mode 100644 sound/soc/codecs/cs35l56-shared.c create mode 100644 sound/soc/codecs/cs35l56-spi.c create mode 100644 sound/soc/codecs/cs35l56.c create mode 100644 sound/soc/codecs/cs35l56.h create mode 100644 sound/soc/intel/boards/sof_sdw_cs35l56.c -- 2.30.2

2 21

Re: Firewire Audio Interface
by Takashi Sakamoto 22 Feb '23

22 Feb '23

Hi, (C.C.ed to alsa-devel) On Tue, Feb 21, 2023 at 10:21:14AM -0500, mj@mj wrote: > Hello Takashi! > I came across your name when trying to find software to make a TC > Konnekt 6 work in OSX Big Sur. > TC has abandoned the Konnekt series. > I saw this Github entry that I think is from you? > [1]https://github.com/alsa-project/snd-firewire-ctl-services/blob/master/REA… > I wanted to ask…is this your project? and would it allow me to use the > Konnekt with OSX 64-bit? I maintain the repository. Linux kernel has driver stack for audio and music unit in IEEE 1394 bus. I maintain the driver stack as well. For example, you can see backport drivers from mainline Linux kernel: * https://github.com/takaswie/snd-firewire-improve/ The repository you found is supplemental one. It provides user space service program to configure DSP functions such as channel strip effects. Besides, they are available in Linux only. As long as I know, no one work to develop craft drivers in MacOS for your device. P.S. I think you use 'Desktop Konnekt 6' or 'Konnekt 8'. Takashi Sakamoto

1 0

[PATCH V3 0/8] Add SoundWire support for AMD platforms
by Vijendar Mukunda 22 Feb '23

22 Feb '23

ACP IP(v6.x) block has two SoundWire manager instance support. This patchset adds support for AMD SoundWire manager driver. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Changes since v2: - Remove useless variable initializations. - Add helper function to interpret peripheral status. - Move runtime pm sequence to probe_work workqueue. - Use string "SoundWire" instead of "soundwire" in code. - Update comments in interrupt handler and probe sequence. - Rename "sdw_lock" as "acp_sdw_lock". - Remove __func__ from dev_dbg statements. Changes since v1: - Drop asoc tree based patches. will send asoc patches as a separate series. - Fixed double space errors. - Use dev instead of pci->dev. - Use SoundWire manager terminology. - Remove amd_sdw_compute_slave_ports() function and use exported sdw_compute_slave_ports() function. - Remove unused variable "num_ports" from amd_manager structure. - Drop startup and shutdown dai callbacks. - Drop reset_page_addr callback. - Use relative address offset to program SoundWire manager registers throughout the code. - Separate wake enable interrupt handling from slave status handling logic. - Use acp_mmio to program ACP common registers. - Use dai_runtime_array implementation in dai_ops. - Refactor port_ops callbacks. - Add comments in port_ops callbacks. - Add retry count logic in irq thread to address faulty case. - Add helper function to interpret command response. - Add generic bandwidth allocation dependency in Kconfig options. - Add comments for AMD SoundWire power modes. - Add missing timeout check in amd_init_sdw_manager callback. - Declare frameshape parameters in probe call. - Handle error case in clock stop sequence. - Add comments in pm_prepare and pm_ops callbacks. Vijendar Mukunda (8): soundwire: export sdw_compute_slave_ports() function soundwire: amd: Add support for AMD Manager driver soundwire: amd: register SoundWire manager dai ops soundwire: amd: enable build for AMD SoundWire manager driver soundwire: amd: add SoundWire manager interrupt handling soundwire: amd: add runtime pm ops for AMD SoundWire manager driver soundwire: amd: handle SoundWire wake enable interrupt soundwire: amd: add pm_prepare callback and pm ops support drivers/soundwire/Kconfig | 10 + drivers/soundwire/Makefile | 4 + drivers/soundwire/amd_manager.c | 1335 +++++++++++++++++ drivers/soundwire/amd_manager.h | 275 ++++ drivers/soundwire/bus.h | 9 + .../soundwire/generic_bandwidth_allocation.c | 12 +- include/linux/soundwire/sdw_amd.h | 108 ++ 7 files changed, 1744 insertions(+), 9 deletions(-) create mode 100644 drivers/soundwire/amd_manager.c create mode 100644 drivers/soundwire/amd_manager.h create mode 100644 include/linux/soundwire/sdw_amd.h -- 2.34.1

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[PATCH 0/2] ASoC: cppcheck fixups
by Kuninori Morimoto 22 Feb '23

22 Feb '23

Hi Mark Cc Takashi This patch-set fixes the issues which is reported by cppcheck. sound/pci/asihpi/hpi6205.c | 2 +- sound/pci/hda/patch_ca0132.c | 4 +++- 2 files changed, 4 insertions(+), 2 deletions(-) It still reports about sound/pci/ice1712/wm8766.c (A) that there is a case that "val2" might be used with uninitialized. But I'm not sure what is the good solution for it. So, I can do is only Reported-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> static int snd_wm8766_ctl_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_wm8766 *wm = snd_kcontrol_chip(kcontrol); int n = kcontrol->private_value; u16 val1, val2; if (wm->ctl[n].get) wm->ctl[n].get(wm, &val1, &val2); else { val1 = wm->regs[wm->ctl[n].reg1] & wm->ctl[n].mask1; val1 >>= __ffs(wm->ctl[n].mask1); if (wm->ctl[n].flags & WM8766_FLAG_STEREO) { val2 = wm->regs[wm->ctl[n].reg2] & wm->ctl[n].mask2; val2 >>= __ffs(wm->ctl[n].mask2); if (wm->ctl[n].flags & WM8766_FLAG_VOL_UPDATE) val2 &= ~WM8766_VOL_UPDATE; } } if (wm->ctl[n].flags & WM8766_FLAG_INVERT) { val1 = wm->ctl[n].max - (val1 - wm->ctl[n].min); if (wm->ctl[n].flags & WM8766_FLAG_STEREO) (A) val2 = wm->ctl[n].max - (val2 - wm->ctl[n].min); } ^^^^ ucontrol->value.integer.value[0] = val1; if (wm->ctl[n].flags & WM8766_FLAG_STEREO) ucontrol->value.integer.value[1] = val2; return 0; } -- 2.25.1

1 2

[PATCH v5 0/4] Add CS35L41 shared boost feature
by Lucas Tanure 21 Feb '23

21 Feb '23

Valve's Steam Deck uses CS35L41 in shared boost mode, where both speakers share the boost circuit. Add this support in the shared lib, but for now, shared boost is not supported in HDA systems as would require BIOS changes. Based on David Rhodes shared boost patches. Also, fix boost config overwriting in IRQ found in the review and do a small refactor of the code. Changes from V4: - Fix Document subject Changes from V3: - Fix wrong code sent - Fix ISO C90 mixed declarations and code Changes from V2: - Drop External boost without VSPK Documentation - Move Shared boost to use values 2 and 3 - Revert back to reg_sequence but reading the value first and only update the necessary bits - Fix bug found by Intel kernel Test Robot Changes from V1: - Fix Documentation patch subject - New patch for External boost without VSPK Documentation - New patch to fix boost IRQ overwriting issue - New patch to refactor IRQ release error code - reinit_completion on pcm_startup - fix DRE switch overwriting - return IRQ_HANDLED in PLL_LOCK case Lucas Tanure (4): ASoC: cs35l41: Only disable internal boost ASoC: cs35l41: Refactor error release code ALSA: cs35l41: Add shared boost feature ASoC: dt-bindings: cirrus,cs35l41: Document CS35l41 shared boost .../bindings/sound/cirrus,cs35l41.yaml | 10 +- include/sound/cs35l41.h | 13 +- sound/pci/hda/cs35l41_hda.c | 6 +- sound/soc/codecs/cs35l41-lib.c | 73 +++++++++- sound/soc/codecs/cs35l41.c | 125 +++++++++--------- sound/soc/codecs/cs35l41.h | 1 + 6 files changed, 157 insertions(+), 71 deletions(-) -- 2.39.1

4 14

[PATCH v2 0/3] ASoC: mchp-pdmc: fix poc noises when starting capture
by Claudiu Beznea 21 Feb '23

21 Feb '23

To start capture on Microchip PDMC the enable bits for each supported microphone need to be set. After this bit is set the PDMC starts to receive data from microphones and it considers this data as valid data. Thus if microphones are not ready the PDMC captures anyway data from its lines. This data is interpreted by the human ear as poc noises. To avoid this the following software workaround need to be applied when starting capture: 1/ enable PDMC channel 2/ wait 150ms 3/ execute 16 dummy reads from RHR 4/ clear interrupts 5/ enable interrupts 6/ enable DMA channel For this workaround to work step 6 need to be executed at the end. For step 6 was added patch 1/3 from this series. With this, component DAI driver sets its struct snd_soc_component_driver::start_dma_last = 1 and proper action is taken based on this flag when starting DAI trigger vs DMA. Thank you, Claudiu Beznea Changes in v2: - patch 1/3 from v1 is now "ASoC: soc-pcm: add option to start DMA after DAI" - pass start_dma_last from component DAI driver object (struct snd_soc_component_driver::start_dma_last); adapt patch 3/3 after this; - in patch 1/3 s/Do we need to start dma first/Do we need to start dma last in comment from soc_pcm_trigger() - collect review tag from Krzysztof Claudiu Beznea (3): ASoC: soc-pcm: add option to start DMA after DAI ASoC: dt-bindings: sama7g5-pdmc: add microchip,startup-delay-us binding ASoC: mchp-pdmc: fix poc noise at capture startup .../sound/microchip,sama7g5-pdmc.yaml | 6 ++ include/sound/soc-component.h | 2 + sound/soc/atmel/mchp-pdmc.c | 55 +++++++++++++++++-- sound/soc/soc-pcm.c | 27 +++++++-- 4 files changed, 80 insertions(+), 10 deletions(-) -- 2.34.1

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alsa-devel mailing list - mailman upgrade
by Jaroslav Kysela 21 Feb '23

21 Feb '23

Hi all, I upgraded the mailman 2 to mailman 3 on the ALSA server. I hope that you do not see any differences, but there is a major change in the handling of the user logins to maintain your subscription using the web interface [1]. It means that you should create a new user account on the mailman server [2]. Do not request the password change for the first login. Although the subscriptions are migrated, the new system will send you a reply that your e-mail is not in the user database. Although it is technically correct, it is very confusing. Create a new account using identical e-mail as you are subscribed. Your subscription will be assigned to this account according your e-mail. You will be allowed to maintain all list subscriptions on the server then (not only one list subscription like before). Archive URLs are different, but I will keep the old archive (URLs), too. Enjoy new Postorius and HyperKitty frontends. If you have any issue, please, let me know. Thanks, Jaroslav [1] https://docs.mailman3.org/en/latest/faq-migration.html [2] https://mailman.alsa-project.org -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

1 0

[PATCH RFC 0/3] Add Versa3 clock generator support
by Biju Das 20 Feb '23

20 Feb '23

The 5P35023 is a VersaClock programmable clock generator and it provides 6 clk outputs {diff2, diff1, se3, se2, se1 and refin}. It has an internal OTP memory allows the user to store the configuration in the device. After power up, the user can change the device register settings through the I2C interface when I2C mode is selected. This driver is for overriding OTP default values during boot based on a full register map from DT, and also minimal support to change the parent of a output clock. The motivation for developing this driver is for supporting 48KHz playback/record with audio codec on RZ/G2L SMARC EVK. On RZ/G2L SMARC EVK, By default audio mclk is connected to 11.2896 MHz clk which is multiple of 44.1KHz. Please see the below default OTP configuration of Dividers connected to output clocks. DIV3 12.2880 MHz DIFF2--> Audio clk2 DIV5 11.2896 MHz SE1 --> Audio clk1 DIV5 11.2896 MHz SE2 --> Audio mck DIV4 12 MHz SE3 --> This clk Not used DIV1 25 MHz DIFF1-->Ethernet clk Ref1-> 24MHz With this setup, we won't be able to do 48KHz playback/record on audio codec, as mck is always connected to 11.2896MHz clk. But by programming the i2c, we can make use of DIV4 to generate 12.2880 MHz and make that as parent of SE2 and there by supporting 48KHz playback/record. A block diagram with modification can be find here[1] [1]https://paste.pics/a253ce7cdc8720c3b5eb6953b97b25ff DIV3 12.2880 MHz DIFF2--> Audio clk2 DIV5 11.2896 MHz SE1 --> Audio clk1 DIV5 11.2896 MHz | SE2 --> Audio mck DIV4 12.2880 MHz | DIV2 12 MHz SE3 --> This clk Not used DIV1 25 MHz DIFF1--> Ethernet clk Ref1-> 24MHz The driver can read a full register map from the DT, and will use that register map to initialize the clk generator when the system boots. and later, based on sampling rate, it switches the parent of SE2 and support both 44.1 and 48 KHz playback/record at run time. 48KHz playback 1f: f6 --> setting Div4 as clock source for se2 Read at address 0x10049C00 : 0x300B4022 --> Setting Audio clk2 in SSI pfd2 1 1 0 24000000 pll2 1 1 0 491519897 div4_mux 1 1 0 491519897 div4 1 1 0 12287998 se2_mux 1 1 0 12287998 se2 1 1 0 12287998 44.1KHz playback 1f: b6 --> setting Div5 as clock source for se2 Read at address 0x10049C00: 0x700B4022--> Setting Audio clk1 in SSI pfd3_mux 1 1 0 24000000 pfd3 1 1 0 960000 pll3 1 1 0 564480000 div5 1 1 0 11289600 se2_mux 1 1 0 11289600 se2 1 1 0 11289600 Please provide your valuable comment for this patch series. Biju Das (3): dt-bindings: clock: Add Renesas versa3 clock generator bindings drivers: clk: Add support for versa3 clock driver arm64: dts: renesas: rzg2l-smarc: Use versa3 clk for audio mclk .../bindings/clock/renesas,versaclock3.yaml | 135 ++ .../boot/dts/renesas/rz-smarc-common.dtsi | 7 - arch/arm64/boot/dts/renesas/rzg2l-smarc.dtsi | 35 + drivers/clk/Kconfig | 9 + drivers/clk/Makefile | 1 + drivers/clk/clk-versaclock3.c | 1134 +++++++++++++++++ 6 files changed, 1314 insertions(+), 7 deletions(-) create mode 100644 Documentation/devicetree/bindings/clock/renesas,versaclock3.yaml create mode 100644 drivers/clk/clk-versaclock3.c -- 2.25.1

1 1

[PATCH BACKPORT 5.4] ASoC: SOF: Intel: hda-dai: fix possible stream_tag leak
by Peter Ujfalusi 20 Feb '23

20 Feb '23

[ Upstream commit 1f810d2b6b2fbdc5279644d8b2c140b1f7c9d43d ] There were just too many code changes since 5.4 stable to prevent clean picking the fix from mainline. The HDaudio stream allocation is done first, and in a second step the LOSIDV parameter is programmed for the multi-link used by a codec. This leads to a possible stream_tag leak, e.g. if a DisplayAudio link is not used. This would happen when a non-Intel graphics card is used and userspace unconditionally uses the Intel Display Audio PCMs without checking if they are connected to a receiver with jack controls. We should first check that there is a valid multi-link entry to configure before allocating a stream_tag. This change aligns the dma_assign and dma_cleanup phases. Cc: stable(a)vger.kernel.org # 5.4.x Complements: b0cd60f3e9f5 ("ALSA/ASoC: hda: clarify bus_get_link() and bus_link_get() helpers") Link: https://github.com/thesofproject/linux/issues/4151 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/20230216162340.19480-1-peter.ujfalusi@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda-dai.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index b3cdd10c83ae..c30e450fa970 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -211,6 +211,10 @@ static int hda_link_hw_params(struct snd_pcm_substream *substream, int stream_tag; int ret; + link = snd_hdac_ext_bus_get_link(bus, codec_dai->component->name); + if (!link) + return -EINVAL; + /* get stored dma data if resuming from system suspend */ link_dev = snd_soc_dai_get_dma_data(dai, substream); if (!link_dev) { @@ -231,10 +235,6 @@ static int hda_link_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - link = snd_hdac_ext_bus_get_link(bus, codec_dai->component->name); - if (!link) - return -EINVAL; - /* set the stream tag in the codec dai dma params */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) snd_soc_dai_set_tdm_slot(codec_dai, stream_tag, 0, 0, 0); -- 2.39.2

2 1

[PATCH BACKPORT 5.10, 5.15] ASoC: SOF: Intel: hda-dai: fix possible stream_tag leak
by Peter Ujfalusi 20 Feb '23

20 Feb '23

[ Upstream commit 1f810d2b6b2fbdc5279644d8b2c140b1f7c9d43d ] There were just too many code changes since 5.10/5.15 stable to prevent clean picking the fix from mainline. The HDaudio stream allocation is done first, and in a second step the LOSIDV parameter is programmed for the multi-link used by a codec. This leads to a possible stream_tag leak, e.g. if a DisplayAudio link is not used. This would happen when a non-Intel graphics card is used and userspace unconditionally uses the Intel Display Audio PCMs without checking if they are connected to a receiver with jack controls. We should first check that there is a valid multi-link entry to configure before allocating a stream_tag. This change aligns the dma_assign and dma_cleanup phases. Cc: stable(a)vger.kernel.org # 5.15.x 5.10.x Complements: b0cd60f3e9f5 ("ALSA/ASoC: hda: clarify bus_get_link() and bus_link_get() helpers") Link: https://github.com/thesofproject/linux/issues/4151 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/20230216162340.19480-1-peter.ujfalusi@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda-dai.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index de80f1b3d7f2..a6275cc92a40 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -212,6 +212,10 @@ static int hda_link_hw_params(struct snd_pcm_substream *substream, int stream_tag; int ret; + link = snd_hdac_ext_bus_get_link(bus, codec_dai->component->name); + if (!link) + return -EINVAL; + /* get stored dma data if resuming from system suspend */ link_dev = snd_soc_dai_get_dma_data(dai, substream); if (!link_dev) { @@ -232,10 +236,6 @@ static int hda_link_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - link = snd_hdac_ext_bus_get_link(bus, codec_dai->component->name); - if (!link) - return -EINVAL; - /* set the hdac_stream in the codec dai */ snd_soc_dai_set_stream(codec_dai, hdac_stream(link_dev), substream->stream); -- 2.39.2

1 0

[PATCH BACKPORT 6.0, 6.1] ASoC: SOF: Intel: hda-dai: fix possible stream_tag leak
by Peter Ujfalusi 20 Feb '23

20 Feb '23

[ Upstream commit 1f810d2b6b2fbdc5279644d8b2c140b1f7c9d43d ] The reason for manual backport is that in 6.2 there were a naming cleanup around hdac, link, hlink, for example snd_hdac_ext_bus_get_link is renamed to snd_hdac_ext_bus_get_hlink_by_name in 6.2. The HDaudio stream allocation is done first, and in a second step the LOSIDV parameter is programmed for the multi-link used by a codec. This leads to a possible stream_tag leak, e.g. if a DisplayAudio link is not used. This would happen when a non-Intel graphics card is used and userspace unconditionally uses the Intel Display Audio PCMs without checking if they are connected to a receiver with jack controls. We should first check that there is a valid multi-link entry to configure before allocating a stream_tag. This change aligns the dma_assign and dma_cleanup phases. Cc: stable(a)vger.kernel.org # 6.1.x 6.0.x Complements: b0cd60f3e9f5 ("ALSA/ASoC: hda: clarify bus_get_link() and bus_link_get() helpers") Link: https://github.com/thesofproject/linux/issues/4151 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/20230216162340.19480-1-peter.ujfalusi@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda-dai.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 556e883a32ed..5f03ee390d54 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -216,6 +216,10 @@ static int hda_link_dma_hw_params(struct snd_pcm_substream *substream, struct hdac_bus *bus = hstream->bus; struct hdac_ext_link *link; + link = snd_hdac_ext_bus_get_link(bus, codec_dai->component->name); + if (!link) + return -EINVAL; + hext_stream = snd_soc_dai_get_dma_data(cpu_dai, substream); if (!hext_stream) { hext_stream = hda_link_stream_assign(bus, substream); @@ -225,10 +229,6 @@ static int hda_link_dma_hw_params(struct snd_pcm_substream *substream, snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)hext_stream); } - link = snd_hdac_ext_bus_get_link(bus, codec_dai->component->name); - if (!link) - return -EINVAL; - /* set the hdac_stream in the codec dai */ snd_soc_dai_set_stream(codec_dai, hdac_stream(hext_stream), substream->stream); -- 2.39.2

1 0

[PATCH] ASoC: Intel: sof_rt5682: Add quirk for Rex board with mx98360a amplifier
by Peter Ujfalusi 20 Feb '23

20 Feb '23

From: "Dharageswari.R" <dharageswari.r(a)intel.com> Add mtl_mx98360a_rt5682 driver data for Chrome Rex board support. Signed-off-by: Dharageswari.R <dharageswari.r(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/intel/boards/sof_rt5682.c | 23 +++++++++++++++++++ .../intel/common/soc-acpi-intel-mtl-match.c | 12 ++++++++++ 2 files changed, 35 insertions(+) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 71a11d747622..4fe448295a90 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -223,6 +223,20 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .callback = sof_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Rex"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98360_ALC5682I_I2S"), + }, + .driver_data = (void *)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(2) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(0) | + SOF_RT5682_NUM_HDMIDEV(4) + ), + }, { .callback = sof_rt5682_quirk_cb, .matches = { @@ -1105,6 +1119,15 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(1) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "mtl_mx98360_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4)), + }, { .name = "jsl_rt5682", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index b1a66a0f6818..7911c3af8071 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -15,6 +15,11 @@ static const struct snd_soc_acpi_codecs mtl_max98357a_amp = { .codecs = {"MX98357A"} }; +static const struct snd_soc_acpi_codecs mtl_max98360a_amp = { + .num_codecs = 1, + .codecs = {"MX98360A"} +}; + static const struct snd_soc_acpi_codecs mtl_rt5682_rt5682s_hp = { .num_codecs = 2, .codecs = {"10EC5682", "RTL5682"}, @@ -28,6 +33,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { .quirk_data = &mtl_max98357a_amp, .sof_tplg_filename = "sof-mtl-max98357a-rt5682.tplg", }, + { + .comp_ids = &mtl_rt5682_rt5682s_hp, + .drv_name = "mtl_mx98360_rt5682", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &mtl_max98360a_amp, + .sof_tplg_filename = "sof-mtl-max98360a-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_mtl_machines); -- 2.39.2

1 0

[PATCH 0/3] SoC: SOF: Intel: hda/mtl: Improve and enable DMI L1 haldling
by Peter Ujfalusi 20 Feb '23

20 Feb '23

Hi, The first patch will improve the managing of DMI L1 by tracking it's enabled/disabled state to avoid unconditional changes to it's state. The remaining two patch will enable the DMI L1 for MTL platforms (ACE 1.0) Regards, Peter --- Ranjani Sridharan (3): ASoC: SOF: Intel: hda: Do not re-enable L1 if disabled before suspend ASoC: SOF: Intel: hda: Restrict DMI L1 disable workaround ASoC: SOF: Intel: MTL: Enable DMI L1 sound/soc/sof/intel/hda-ctrl.c | 8 +++++--- sound/soc/sof/intel/hda-dsp.c | 10 ++++------ sound/soc/sof/intel/hda-stream.c | 15 ++++++++++++--- sound/soc/sof/intel/hda.h | 2 +- sound/soc/sof/intel/mtl.c | 3 +++ sound/soc/sof/intel/mtl.h | 2 ++ 6 files changed, 27 insertions(+), 13 deletions(-) -- 2.39.2

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[regression] Bug 216909 - Avantree Leaf USB audio device does not play sound in applications run via wine
by Linux kernel regression tracking (Thorsten Leemhuis) 19 Feb '23

19 Feb '23

Hi, this is your Linux kernel regression tracker. I noticed a regression report in bugzilla.kernel.org. As many (most?) kernel developer don't keep an eye on it, I decided to forward it by mail. Quoting from https://bugzilla.kernel.org/show_bug.cgi?id=216909 : > Bernd Steinhauser 2023-01-10 14:45:18 UTC > > Created attachment 303568 [details] > lsusb -v -d 0a12:1004 > > On my second PC, I'm using a USB audio device called "Avantree Leaf" to pass audio via bluetooth to a receiver. About 1-2 months ago, I noticed that I have no audio with this device when playing sounds via wine applications (e.g. foobar2000 or any kind of game via proton/wine), neither via the pulseaudio output, nor alsa. > Native Linux applications work fine (or at least I haven't found one yet that doesn't work) via both pulse or alsa output. > If I select another audio device (e.g. the speakers of the screen via HDMI) in winecfg for the respective wine prefix, then the audio does work. > Also, if I "prime" the device by playing audio via a native application and then the audio works. > > I'm using pipewire-0.3.63 with pipewire-pulse as an audio server, but I've also tried with recent pulseaudio and they are both affected. > For pipewire, I played a bit with the settings and found: > Brings back audio: > clock.min-quantum 512 (lower values don't work) > > Doesn't change the outcome: > ["api.alsa.period-size"] = 16 up to 1024 > ["api.alsa.disable-batch"] = true/false (in combination with the above) > ["api.alsa.headroom"] = 64 up to 1024 > > The output I get from `cat /proc/asound/Leaf/pcm0p/sub0/hw_params`: > Native application: > access: MMAP_INTERLEAVED > format: S16_LE > subformat: STD > channels: 2 > rate: 48000 (48000/1) > period_size: 256 > buffer_size: 32768 > > foobar2000 via wine: > access: MMAP_INTERLEAVED > format: S16_LE > subformat: STD > channels: 2 > rate: 48000 (48000/1) > period_size: 128 > buffer_size: 32768 > > So they are the same, except for the period_size. > > Yesterday, I finally made some effort to track this down and I had to go back all the way to v5.14 to find the commit that is responsible: > https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?… > > If I test the revision before that commit, everything works as expected. If I test this revision, then the audio fails. > Of course with such an old commit, I'm not 100% sure that this is the only thing, as I can't simply revert it on a newer kernel. Too much change happened in sound/usb since then. > I did however test every mainline kernel version since 5.10 and the behavior was consistent. > Please let me know if you need any further information. See the ticket for more details. BTW, let me use this mail to also add the report to the list of tracked regressions to ensure it's doesn't fall through the cracks: #regzbot introduced: 307cc9baac5c07b23d https://bugzilla.kernel.org/show_bug.cgi?id=216909 #regzbot title: sound: avantree Leaf USB audio device stopped playing sound in applications run via wine #regzbot ignore-activity This isn't a regression? This issue or a fix for it are already discussed somewhere else? It was fixed already? You want to clarify when the regression started to happen? Or point out I got the title or something else totally wrong? Then just reply and tell me -- ideally while also telling regzbot about it, as explained by the page listed in the footer of this mail. Developers: When fixing the issue, remember to add 'Link:' tags pointing to the report (e.g. the buzgzilla ticket and maybe this mail as well, if this thread sees some discussion). See page linked in footer for details. Ciao, Thorsten (wearing his 'the Linux kernel's regression tracker' hat) -- Everything you wanna know about Linux kernel regression tracking: https://linux-regtracking.leemhuis.info/about/#tldr If I did something stupid, please tell me, as explained on that page.

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[PATCH] Input: mainstone: fix pxa2xx_ac97 dependency
by Arnd Bergmann 17 Feb '23

17 Feb '23

From: Arnd Bergmann <arnd(a)arndb.de> The exported PXA AC97 symbols are provided by the snd-pxa2xx-lib module if SND_PXA2XX_SOC_AC97 is set, so both symbols have to be enabled for the mainstone driver to actually work, and if snd-pxa2xx-lib is a loadable module, the touchscreen driver cannot be built-in: arm-linux-gnueabi-ld: drivers/input/touchscreen/mainstone-wm97xx.o: in function `wm97xx_acc_pen_up': mainstone-wm97xx.c:(.text+0x8c): undefined reference to `pxa2xx_ac97_read_modr' arm-linux-gnueabi-ld: drivers/input/touchscreen/mainstone-wm97xx.o: in function `wm97xx_acc_pen_down': mainstone-wm97xx.c:(.text+0x2e0): undefined reference to `pxa2xx_ac97_read_modr' The misconfiguration has always been possible but I never hit this after my previous incomplete fix in 2019: since there were many options that each 'select SND_PXA2XX_SOC_AC97', randconfig builds for PXA would almost always have this built-in, not in a loadable module. Fixes: e217b085a1ac ("Input: touchscreen: use wrapper for pxa2xx ac97 registers") Fixes: b401d1fd8053 ("ASoC: pxa: remove unused board support") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- drivers/input/touchscreen/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/input/touchscreen/Kconfig b/drivers/input/touchscreen/Kconfig index 61bb8e632413..1601f5bae005 100644 --- a/drivers/input/touchscreen/Kconfig +++ b/drivers/input/touchscreen/Kconfig @@ -903,7 +903,7 @@ config TOUCHSCREEN_WM9713 config TOUCHSCREEN_WM97XX_MAINSTONE tristate "WM97xx Mainstone/Palm accelerated touch" depends on TOUCHSCREEN_WM97XX && ARCH_PXA - depends on SND_PXA2XX_LIB_AC97 + depends on SND_PXA2XX_LIB && SND_PXA2XX_LIB_AC97 help Say Y here for support for streaming mode with WM97xx touchscreens on Mainstone, Palm Tungsten T5, TX and LifeDrive systems. -- 2.39.1

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[PATCH v5 00/10] Add the PowerQUICC audio support using the QMC
by Herve Codina 17 Feb '23

17 Feb '23

Hi, This series adds support for audio using the QMC controller available in some Freescale PowerQUICC SoCs. This series contains three parts in order to show the different blocks hierarchy and their usage in this support. The first one is related to TSA (Time Slot Assigner). The TSA handles the data present at the pin level (TDM with up to 64 time slots) and dispatchs them to one or more serial controller (SCC). The second is related to QMC (QUICC Multichannel Controller). The QMC handles the data at the serial controller (SCC) level and splits again the data to creates some virtual channels. The last one is related to the audio component (QMC audio). It is the glue between the QMC controller and the ASoC component. It handles one or more QMC virtual channels and creates one DAI per QMC virtual channels handled. Compared to the previous iteration https://lore.kernel.org/linux-kernel/20230126083222.374243-1-herve.codina@b… this v5 series mainly: - fixes bindings, - removes one left out_8() specific ppc call (missed in v3), - changes 'depends-on' in case of COMPILE_TEST. Best regards, Herve Codina Changes v4 -> v5 - patch 1 Rename fsl,tsa.yaml to fsl,cpm1-tsa.yaml Rename #serial-cells to #fsl,serial-cells and add a description Fix typos Remove examples present in description Use a pattern property for fsl,[rt]x-ts-routes - patch 2 Remove one left out_8() ppc specific function call Remove the no more needed PPC dependency in case of COMPILE_TEST - patch 4 Add 'Acked-by: Michael Ellerman <mpe(a)ellerman.id.au>' - patch 5 Rename fsl,qmc.yaml to fsl,cpm1-scc-qmc.yaml Rename #chan-cells to #fsl,chan-cells and add a description - patch 6 Add the SOC_FSL dependency in case of COMPILE_TEST (issue raised by the kernel test robot). Fix a typo in commit log Add 'Acked-by: Li Yang <leoyang.li(a)nxp.com>' Changes v3 -> v4 - patches 2, 6 and 9 Update code comment format. - patch 1 Fix some description formats. Add 'additionalProperties: false' in subnode. Move fsl,mode to fsl,diagnostic-mode. Change clocks and clock-names properties. Add '#serial-cells' property related to the newly introduced fsl,tsa-serial phandle. - patch 2 Move fsl,mode to fsl,diagnostic-mode. Replace the fsl,tsa phandle and the fsl,tsa-cell-id property by a fsl,tsa-serial phandle and update the related API. Add missing locks. - patch 5 Fix some description format. Replace the fsl,tsa phandle and the fsl,tsa-cell-id property by a fsl,tsa-serial phandle. Rename fsl,mode to fsl,operational-mode and update its description. - patch 6 Replace the fsl,tsa phandle and the fsl,tsa-cell-id property by a fsl,tsa-serial phandle and use the TSA updated API. Rename fsl,mode to fsl,operational-mode. - patch 8 Add 'Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>' Changes v2 -> v3 - All bindings Rename fsl-tsa.h to fsl,tsa.h Add missing vendor prefix Various fixes (quotes, node names, upper/lower case) - patches 1 and 2 (TSA binding specific) Remove 'reserved' values in the routing tables Remove fsl,grant-mode Add a better description for 'fsl,common-rxtx-pins' Fix clocks/clocks-name handling against fsl,common-rxtx-pins Add information related to the delays unit Removed FSL_CPM_TSA_NBCELL Fix license in binding header file fsl,tsa.h - patches 5 and 6 (QMC binding specific) Remove fsl,cpm-command property Add interrupt property constraint - patches 8 and 9 (QMC audio binding specific) Remove 'items' in compatible property definition Add missing 'dai-common.yaml' reference Fix the qmc_chan phandle definition - patch 2 and 6 Use io{read,write}be{32,16} Change commit subjects and logs - patch 4 Add 'Acked-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' Changes v1 -> v2: - patch 2 and 6 Fix kernel test robot errors - other patches No changes Herve Codina (10): dt-bindings: soc: fsl: cpm_qe: Add TSA controller soc: fsl: cpm1: Add support for TSA MAINTAINERS: add the Freescale TSA controller entry powerpc/8xx: Use a larger CPM1 command check mask dt-bindings: soc: fsl: cpm_qe: Add QMC controller soc: fsl: cpm1: Add support for QMC MAINTAINERS: add the Freescale QMC controller entry dt-bindings: sound: Add support for QMC audio ASoC: fsl: Add support for QMC audio MAINTAINERS: add the Freescale QMC audio entry .../soc/fsl/cpm_qe/fsl,cpm1-scc-qmc.yaml | 172 ++ .../bindings/soc/fsl/cpm_qe/fsl,cpm1-tsa.yaml | 234 +++ .../bindings/sound/fsl,qmc-audio.yaml | 117 ++ MAINTAINERS | 25 + arch/powerpc/platforms/8xx/cpm1.c | 2 +- drivers/soc/fsl/qe/Kconfig | 23 + drivers/soc/fsl/qe/Makefile | 2 + drivers/soc/fsl/qe/qmc.c | 1533 +++++++++++++++++ drivers/soc/fsl/qe/tsa.c | 869 ++++++++++ drivers/soc/fsl/qe/tsa.h | 42 + include/dt-bindings/soc/fsl,tsa.h | 13 + include/soc/fsl/qe/qmc.h | 71 + sound/soc/fsl/Kconfig | 9 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_qmc_audio.c | 735 ++++++++ 15 files changed, 3848 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/soc/fsl/cpm_qe/fsl,cpm1-scc-qmc.yaml create mode 100644 Documentation/devicetree/bindings/soc/fsl/cpm_qe/fsl,cpm1-tsa.yaml create mode 100644 Documentation/devicetree/bindings/sound/fsl,qmc-audio.yaml create mode 100644 drivers/soc/fsl/qe/qmc.c create mode 100644 drivers/soc/fsl/qe/tsa.c create mode 100644 drivers/soc/fsl/qe/tsa.h create mode 100644 include/dt-bindings/soc/fsl,tsa.h create mode 100644 include/soc/fsl/qe/qmc.h create mode 100644 sound/soc/fsl/fsl_qmc_audio.c -- 2.39.1

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[PATCH v3] ASoC: tas2781: Add tas2781 driver
by Shenghao Ding 17 Feb '23

17 Feb '23

Create tas2781 driver. Signed-off-by: Shenghao Ding <13916275206(a)139.com> --- Changes in v3: - fixed issues reported-by: kernel test robot <lkp(a)intel.com> - correct all the inconsistent indenting warnings. - correct uninitialized symbol 'pwr_state' - correct unsigned 'val' is never less than zero - correct 'tas_dev' could be null Changes to be committed: modified: sound/soc/codecs/Kconfig modified: sound/soc/codecs/Makefile new file: sound/soc/codecs/tas2781-dsp.c new file: sound/soc/codecs/tas2781-dsp.h new file: sound/soc/codecs/tas2781-i2c.c new file: sound/soc/codecs/tas2781.h --- sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2781-dsp.c | 2253 ++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2781-dsp.h | 178 +++ sound/soc/codecs/tas2781-i2c.c | 1913 +++++++++++++++++++++++++++ sound/soc/codecs/tas2781.h | 176 +++ 6 files changed, 4534 insertions(+) create mode 100644 sound/soc/codecs/tas2781-dsp.c create mode 100644 sound/soc/codecs/tas2781-dsp.h create mode 100644 sound/soc/codecs/tas2781-i2c.c create mode 100644 sound/soc/codecs/tas2781.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0f9d71490075..9016c4846d95 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -223,6 +223,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2764 imply SND_SOC_TAS2770 imply SND_SOC_TAS2780 + imply SND_SOC_TAS2781 imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -1579,6 +1580,17 @@ config SND_SOC_TAS2780 Enable support for Texas Instruments TAS2780 high-efficiency digital input mono Class-D audio power amplifiers. +config SND_SOC_TAS2781 + tristate "Texas Instruments TAS2781 speaker amplifier" + depends on I2C + select REGMAP_I2C + select CRC8 + help + Enable support for Texas Instruments TAS2781 Smart Amplifier + Digital input mono Class-D and DSP-inside audio power amplifiers. + Note the TAS2781 driver implements a flexible and configurable + algo coff setting, for one, two, even multiple TAS2781 chips. + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 71d3ce5867e4..49f912aea19d 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -355,6 +355,7 @@ snd-soc-tas2552-objs := tas2552.o snd-soc-tas2562-objs := tas2562.o snd-soc-tas2764-objs := tas2764.o snd-soc-tas2780-objs := tas2780.o +snd-soc-tas2781-objs := tas2781-i2c.o tas2781-dsp.o # Mux snd-soc-simple-mux-objs := simple-mux.o @@ -606,6 +607,7 @@ obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o obj-$(CONFIG_SND_SOC_TAS2562) += snd-soc-tas2562.o obj-$(CONFIG_SND_SOC_TAS2764) += snd-soc-tas2764.o obj-$(CONFIG_SND_SOC_TAS2780) += snd-soc-tas2780.o +obj-$(CONFIG_SND_SOC_TAS2781) += snd-soc-tas2781.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2781-dsp.c b/sound/soc/codecs/tas2781-dsp.c new file mode 100644 index 000000000000..08c554481ff0 --- /dev/null +++ b/sound/soc/codecs/tas2781-dsp.c @@ -0,0 +1,2253 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier + * + * Copyright (C) 2022 - 2023 Texas Instruments Incorporated + * https://www.ti.com + * + * The TAS2781 driver implements a flexible and configurable algo coff setting + * for one, two, even multiple TAS2781 chips. + * + * Author: Shenghao Ding <shenghao-ding(a)ti.com> + * Kevin Lu <kevin-lu(a)ti.com> + */ + +#include <linux/regmap.h> +#include <linux/i2c.h> +#include <linux/string.h> +#include <linux/crc8.h> +#include <linux/slab.h> +#include <linux/fs.h> +#include <linux/firmware.h> + +#include "tas2781.h" + +#define ERROR_PRAM_CRCCHK 0x0000000 +#define ERROR_YRAM_CRCCHK 0x0000001 +#define BINFILE_VER 0 +#define DRV_VER 1 +#define PPC_DRIVER_CRCCHK 0x00000200 + +#define TAS2781_SA_COEFF_SWAP_REG TASDEVICE_REG(0, 0x35, 0x2c) +#define TAS2781_YRAM_BOOK1 140 +#define TAS2781_YRAM1_PAGE 42 +#define TAS2781_YRAM1_START_REG 88 + +#define TAS2781_YRAM2_START_PAGE 43 +#define TAS2781_YRAM2_END_PAGE 49 +#define TAS2781_YRAM2_START_REG 8 +#define TAS2781_YRAM2_END_REG 127 + +#define TAS2781_YRAM3_PAGE 50 +#define TAS2781_YRAM3_START_REG 8 +#define TAS2781_YRAM3_END_REG 27 + +/*should not include B0_P53_R44-R47 */ +#define TAS2781_YRAM_BOOK2 0 +#define TAS2781_YRAM4_START_PAGE 50 +#define TAS2781_YRAM4_END_PAGE 60 + +#define TAS2781_YRAM5_PAGE 61 +#define TAS2781_YRAM5_START_REG 8 +#define TAS2781_YRAM5_END_REG 27 + +#define TASDEVICE_MAXPROGRAM_NUM_KERNEL 5 +#define TASDEVICE_MAXCONFIG_NUM_KERNEL_MULTIPLE_AMPS 64 +#define TASDEVICE_MAXCONFIG_NUM_KERNEL 10 +#define MAIN_ALL_DEVICES_1X 0x01 +#define MAIN_DEVICE_A_1X 0x02 +#define MAIN_DEVICE_B_1X 0x03 +#define MAIN_DEVICE_C_1X 0x04 +#define MAIN_DEVICE_D_1X 0x05 +#define COEFF_DEVICE_A_1X 0x12 +#define COEFF_DEVICE_B_1X 0x13 +#define COEFF_DEVICE_C_1X 0x14 +#define COEFF_DEVICE_D_1X 0x15 +#define PRE_DEVICE_A_1X 0x22 +#define PRE_DEVICE_B_1X 0x23 +#define PRE_DEVICE_C_1X 0x24 +#define PRE_DEVICE_D_1X 0x25 + +struct tas_crc { + unsigned char offset; + unsigned char len; +}; + +const char deviceNumber[TASDEVICE_DSP_TAS_MAX_DEVICE] = { + 1, 2, 1, 2, 1, 1, 0, 2, 4, 3, 1, 2, 3, 4 +}; + +static int fw_parse_block_data_kernel(struct tasdevice_fw *tas_fmw, + struct tasdev_blk *block, const struct firmware *fmw, int offset) +{ + const unsigned char *data = fmw->data; + + if (offset + 4 > fmw->size) { + dev_err(tas_fmw->dev, "%s: File Size error\n", __func__); + offset = -1; + goto out; + } + block->type = SMS_HTONL(data[offset], + data[offset + 1], data[offset + 2], + data[offset + 3]); + offset += 4; + + if (offset + 1 > fmw->size) { + dev_err(tas_fmw->dev, "%s: mbPChkSumPresent error\n", + __func__); + offset = -1; + goto out; + } + block->is_pchksum_present = data[offset]; + offset++; + + if (offset + 1 > fmw->size) { + dev_err(tas_fmw->dev, "%s: mnPChkSum error\n", __func__); + offset = -1; + goto out; + } + block->pchksum = data[offset]; + offset++; + + if (offset + 1 > fmw->size) { + dev_err(tas_fmw->dev, "%s: mbYChkSumPresent error\n", + __func__); + offset = -1; + goto out; + } + block->is_ychksum_present = data[offset]; + offset++; + + if (offset + 1 > fmw->size) { + dev_err(tas_fmw->dev, "%s: mnYChkSum error\n", __func__); + offset = -1; + goto out; + } + block->ychksum = data[offset]; + offset++; + + if (offset + 4 > fmw->size) { + dev_err(tas_fmw->dev, "%s: blk_size error\n", __func__); + offset = -1; + goto out; + } + block->blk_size = SMS_HTONL(data[offset], data[offset + 1], + data[offset + 2], data[offset + 3]); + offset += 4; + + if (offset + 4 > fmw->size) { + dev_err(tas_fmw->dev, "%s: nSublocks error\n", __func__); + offset = -1; + goto out; + } + block->n_subblocks = SMS_HTONL(data[offset], data[offset + 1], + data[offset + 2], data[offset + 3]); + offset += 4; + + block->data = kzalloc(block->blk_size, GFP_KERNEL); + if (block->data == NULL) { + offset = -1; + goto out; + } + memcpy(block->data, &data[offset], block->blk_size); + offset += block->blk_size; +out: + return offset; +} + +static int fw_parse_data_kernel(struct tasdevice_fw *tas_fmw, + struct tasdevice_data *img_data, const struct firmware *fmw, + int offset) +{ + const unsigned char *data = fmw->data; + unsigned int n_block; + + if (offset + 4 > fmw->size) { + dev_err(tas_fmw->dev, "%s: File Size error\n", __func__); + offset = -1; + goto out; + } + img_data->n_blk = SMS_HTONL(data[offset], + data[offset + 1], data[offset + 2], data[offset + 3]); + offset += 4; + + img_data->dev_blks = + kcalloc(img_data->n_blk, sizeof(struct tasdev_blk), + GFP_KERNEL); + if (img_data->dev_blks == NULL) { + dev_err(tas_fmw->dev, "%s: FW memory failed!\n", __func__); + goto out; + } + + for (n_block = 0; n_block < img_data->n_blk; n_block++) { + offset = fw_parse_block_data_kernel(tas_fmw, + &(img_data->dev_blks[n_block]), fmw, offset); + if (offset < 0) { + offset = -1; + goto out; + } + } +out: + return offset; +} + +static int fw_parse_program_data_kernel( + struct tasdevice_priv *tas_dev, struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset) +{ + struct tasdevice_prog *program; + unsigned int n_program = 0; + + for (n_program = 0; n_program < tas_fmw->n_programs; n_program++) { + program = &(tas_fmw->programs[n_program]); + if (offset + 64 > fmw->size) { + dev_err(tas_dev->dev, "%s: mpName error\n", + __func__); + offset = -1; + goto out; + } + offset += 64; + + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnAppMode error\n", + __func__); + offset = -1; + goto out; + } + offset++; + + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnPDMI2SMode error\n", + __func__); + offset = -1; + goto out; + } + offset++; + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnISnsPD error\n", + __func__); + offset = -1; + goto out; + } + offset++; + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnVSnsPD error\n", + __func__); + offset = -1; + goto out; + } + offset++; + //skip 3-byte reserved + offset += 3; + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnPowerLDG error\n", + __func__); + offset = -1; + goto out; + } + offset++; + + offset = fw_parse_data_kernel(tas_fmw, &(program->dev_data), + fmw, offset); + if (offset < 0) + goto out; + } +out: + return offset; +} + +static int fw_parse_configuration_data_kernel( + struct tasdevice_priv *tas_dev, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + const unsigned char *data = fmw->data; + unsigned int n_configs; + struct tasdevice_config *config; + + for (n_configs = 0; n_configs < tas_fmw->n_configurations; + n_configs++) { + config = + &(tas_fmw->configs[n_configs]); + if (offset + 64 > fmw->size) { + dev_err(tas_dev->dev, "%s: mpName error\n", + __func__); + offset = -1; + goto out; + } + memcpy(config->name, &data[offset], 64); + offset += 64; + + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, "%s: orientation error\n", + __func__); + offset = -1; + goto out; + } + offset++; + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnDevices error\n", + __func__); + offset = -1; + goto out; + } + offset += 1; + + if (offset + 2 > fmw->size) { + dev_err(tas_dev->dev, "%s: mProgram error\n", + __func__); + offset = -1; + goto out; + } + offset += 2; + + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnSamplingRate error\n", + __func__); + offset = -1; + goto out; + } + offset += 4; + + if (offset + 2 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnPLLSrc error\n", + __func__); + offset = -1; + goto out; + } + offset += 2; + + if (offset + 2 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnFsRate error\n", + __func__); + offset = -1; + goto out; + } + offset += 2; + + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnPLLSrcRate error\n", + __func__); + offset = -1; + goto out; + } + offset += 4; + offset = fw_parse_data_kernel(tas_fmw, + &(config->dev_data), fmw, offset); + if (offset < 0) + goto out; + } +out: + return offset; +} + +static int fw_parse_variable_header_kernel( + struct tasdevice_priv *tas_dev, const struct firmware *fmw, int offset) +{ + struct tasdevice_fw *tas_fmw = tas_dev->fmw; + struct tasdevice_dspfw_hdr *fw_hdr = &(tas_fmw->fw_hdr); + const unsigned char *buf = fmw->data; + struct tasdevice_prog *program; + struct tasdevice_config *config; + unsigned int n_program = 0, n_configs = 0; + unsigned short max_confs = TASDEVICE_MAXCONFIG_NUM_KERNEL; + + if (offset + 2 > fmw->size) { + dev_err(tas_dev->dev, "%s: File Size error\n", __func__); + offset = -1; + goto out; + } + fw_hdr->device_family = SMS_HTONS(buf[offset], buf[offset + 1]); + if (fw_hdr->device_family != 0) { + dev_err(tas_dev->dev, "ERROR:%s:not TAS device\n", __func__); + offset = -1; + goto out; + } + offset += 2; + if (offset + 2 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnDevice error\n", __func__); + offset = -1; + goto out; + } + fw_hdr->device = SMS_HTONS(buf[offset], buf[offset + 1]); + if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || + fw_hdr->device == 6) { + dev_err(tas_dev->dev, "ERROR:%s: not support device %d\n", + __func__, fw_hdr->device); + offset = -1; + goto out; + } + offset += 2; + fw_hdr->ndev = deviceNumber[fw_hdr->device]; + + if (fw_hdr->ndev != tas_dev->ndev) { + dev_err(tas_dev->dev, + "%s: ndev(%u) in dspbin dismatch ndev(%u) in DTS\n", + __func__, fw_hdr->ndev, tas_dev->ndev); + offset = -1; + goto out; + } + + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnPrograms error\n", __func__); + offset = -1; + goto out; + } + tas_fmw->n_programs = SMS_HTONL(buf[offset], buf[offset + 1], + buf[offset + 2], buf[offset + 3]); + offset += 4; + + if (tas_fmw->n_programs == 0 || tas_fmw->n_programs > + TASDEVICE_MAXPROGRAM_NUM_KERNEL) { + dev_err(tas_dev->dev, "%s: mnPrograms is invalid\n", __func__); + offset = -1; + goto out; + } + + if (offset + 4 * TASDEVICE_MAXPROGRAM_NUM_KERNEL > fmw->size) { + dev_err(tas_dev->dev, "%s: mpPrograms error\n", __func__); + offset = -1; + goto out; + } + + tas_fmw->programs = + kcalloc(tas_fmw->n_programs, + sizeof(struct tasdevice_prog), GFP_KERNEL); + if (tas_fmw->programs == NULL) { + dev_err(tas_dev->dev, "%s: mpPrograms memory failed!\n", + __func__); + offset = -1; + goto out; + } + + for (n_program = 0; n_program < tas_fmw->n_programs; n_program++) { + program = &(tas_fmw->programs[n_program]); + program->prog_size = SMS_HTONL(buf[offset], buf[offset + 1], + buf[offset + 2], buf[offset + 3]); + offset += 4; + } + offset += (4 * (5 - n_program)); + + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnConfigurations error\n", + __func__); + offset = -1; + goto out; + } + tas_fmw->n_configurations = SMS_HTONL(buf[offset], buf[offset + 1], + buf[offset + 2], buf[offset + 3]); + offset += 4; + max_confs = (fw_hdr->ndev >= 4) ? + TASDEVICE_MAXCONFIG_NUM_KERNEL_MULTIPLE_AMPS : + TASDEVICE_MAXCONFIG_NUM_KERNEL; + if (tas_fmw->n_configurations == 0 || + tas_fmw->n_configurations > max_confs) { + dev_err(tas_dev->dev, "%s: mnConfigurations is invalid\n", + __func__); + offset = -1; + goto out; + } + + if (offset + 4 * max_confs > fmw->size) { + dev_err(tas_dev->dev, "%s: mpConfigurations error\n", + __func__); + offset = -1; + goto out; + } + + tas_fmw->configs = kcalloc(tas_fmw->n_configurations, + sizeof(struct tasdevice_config), GFP_KERNEL); + if (tas_fmw->configs == NULL) { + offset = -1; + goto out; + } + + for (n_configs = 0; n_configs < tas_fmw->n_programs; + n_configs++) { + config = + &(tas_fmw->configs[n_configs]); + config->cfg_size = SMS_HTONL(buf[offset], + buf[offset + 1], buf[offset + 2], buf[offset + 3]); + offset += 4; + } + + offset += (4 * (max_confs - n_configs)); +out: + return offset; +} + +static int tasdevice_load_block_kernel( + struct tasdevice_priv *tasdevice, struct tasdev_blk *block) +{ + int ret = 0; + + unsigned char *data = block->data; + unsigned int i = 0, length = 0; + const unsigned int blk_size = block->blk_size; + unsigned char dev_idx = 0; + struct tasdevice_dspfw_hdr *fw_hdr = &(tasdevice->fmw->fw_hdr); + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr = &(fw_hdr->fixed_hdr); + + if (fw_fixed_hdr->ppcver >= PPC3_VERSION) { + switch (block->type) { + case MAIN_ALL_DEVICES_1X: + dev_idx = 0|0x80; + break; + case MAIN_DEVICE_A_1X: + dev_idx = 1|0x80; + break; + case COEFF_DEVICE_A_1X: + case PRE_DEVICE_A_1X: + dev_idx = 1|0xC0; + break; + case MAIN_DEVICE_B_1X: + dev_idx = 2|0x80; + break; + case COEFF_DEVICE_B_1X: + case PRE_DEVICE_B_1X: + dev_idx = 2|0xC0; + break; + case MAIN_DEVICE_C_1X: + dev_idx = 3|0x80; + break; + case COEFF_DEVICE_C_1X: + case PRE_DEVICE_C_1X: + dev_idx = 3|0xC0; + break; + case MAIN_DEVICE_D_1X: + dev_idx = 4|0x80; + break; + case COEFF_DEVICE_D_1X: + case PRE_DEVICE_D_1X: + dev_idx = 4|0xC0; + break; + default: + dev_info(tasdevice->dev, + "%s: load block: Other Type = 0x%02x\n", + __func__, block->type); + break; + } + } else { + switch (block->type) { + case MAIN_ALL_DEVICES: + dev_idx = 0|0x80; + break; + case MAIN_DEVICE_A: + dev_idx = 1|0x80; + break; + case COEFF_DEVICE_A: + case PRE_DEVICE_A: + dev_idx = 1|0xC0; + break; + case MAIN_DEVICE_B: + dev_idx = 2|0x80; + break; + case COEFF_DEVICE_B: + case PRE_DEVICE_B: + dev_idx = 2|0xC0; + break; + case MAIN_DEVICE_C: + dev_idx = 3|0x80; + break; + case COEFF_DEVICE_C: + case PRE_DEVICE_C: + dev_idx = 3|0xC0; + break; + case MAIN_DEVICE_D: + dev_idx = 4|0x80; + break; + case COEFF_DEVICE_D: + case PRE_DEVICE_D: + dev_idx = 4|0xC0; + break; + default: + dev_info(tasdevice->dev, + "%s: load block: Other Type = 0x%02x\n", + __func__, block->type); + break; + } + } + + for (i = 0; i < block->n_subblocks; i++) { + int rc = tasdevice_process_block(tasdevice, data + length, + dev_idx, blk_size - length); + if (rc < 0) { + dev_err(tasdevice->dev, + "%s: ERROR:%u %u sublock write error\n", + __func__, length, blk_size); + break; + } + length += (unsigned int)rc; + if (blk_size < length) { + dev_err(tasdevice->dev, + "%s: ERROR:%u %u out of memory\n", + __func__, length, blk_size); + break; + } + } + + return ret; +} + +static int fw_parse_variable_header_git(struct tasdevice_priv *tas_dev, + const struct firmware *fmw, int offset) +{ + const unsigned char *buf = fmw->data; + struct tasdevice_fw *tas_fmw = tas_dev->fmw; + struct tasdevice_dspfw_hdr *fw_hdr = &(tas_fmw->fw_hdr); + int i = strlen((char *)&buf[offset]); + + i++; + + if (offset + i > fmw->size) { + dev_err(tas_dev->dev, "%s: File Size error\n", __func__); + offset = -1; + goto out; + } + + offset += i; + + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: File Size error\n", __func__); + offset = -1; + goto out; + } + fw_hdr->device_family = SMS_HTONL(buf[offset], buf[offset + 1], + buf[offset + 2], buf[offset + 3]); + if (fw_hdr->device_family != 0) { + dev_err(tas_dev->dev, "ERROR:%s: not TAS device\n", __func__); + offset = -1; + goto out; + } + offset += 4; + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: File Size error\n", __func__); + offset = -1; + goto out; + } + fw_hdr->device = SMS_HTONL(buf[offset], buf[offset + 1], + buf[offset + 2], buf[offset + 3]); + if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || + fw_hdr->device == 6) { + dev_err(tas_dev->dev, "ERROR:%s: not support device %d\n", + __func__, fw_hdr->device); + offset = -1; + goto out; + } + offset += 4; + fw_hdr->ndev = deviceNumber[fw_hdr->device]; + if (fw_hdr->ndev != tas_dev->ndev) { + dev_err(tas_dev->dev, + "%s: ndev(%u) in dspbin dismatch ndev(%u) in DTS\n", + __func__, fw_hdr->ndev, + tas_dev->ndev); + offset = -1; + } + +out: + return offset; +} + +static int fw_parse_variable_hdr_cal(struct tasdevice_priv *tas_dev, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + const unsigned char *buf = fmw->data; + struct tasdevice_dspfw_hdr *fw_hdr = &(tas_fmw->fw_hdr); + int i = strlen((char *)&buf[offset]); + + i++; + + if (offset + i > fmw->size) { + dev_err(tas_dev->dev, "%s: File Size error\n", __func__); + offset = -1; + goto out; + } + + offset += i; + + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnDeviceFamily error\n", __func__); + offset = -1; + goto out; + } + fw_hdr->device_family = SMS_HTONL(buf[offset], buf[offset + 1], + buf[offset + 2], buf[offset + 3]); + if (fw_hdr->device_family != 0) { + dev_err(tas_dev->dev, "ERROR:%s: not TAS device\n", __func__); + offset = -1; + goto out; + } + offset += 4; + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnDevice error\n", __func__); + offset = -1; + goto out; + } + fw_hdr->device = SMS_HTONL(buf[offset], buf[offset + 1], + buf[offset + 2], buf[offset + 3]); + if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || + fw_hdr->device == 6) { + dev_err(tas_dev->dev, "ERROR:%s: not support device %d\n", + __func__, fw_hdr->device); + offset = -1; + goto out; + } + offset += 4; + fw_hdr->ndev = deviceNumber[fw_hdr->device]; + if (fw_hdr->ndev != 1) { + dev_err(tas_dev->dev, + "%s: calbin must be 1, but currently ndev(%u)\n", + __func__, fw_hdr->ndev); + offset = -1; + } + +out: + return offset; +} + +static inline void tas2781_clear_calfirmware(struct tasdevice_fw + *tas_fmw) +{ + int i = 0; + unsigned int n_blk = 0; + + if (tas_fmw->calibrations) { + struct tasdevice_calibration *calibration; + + for (i = 0; i < tas_fmw->n_calibrations; i++) { + calibration = &(tas_fmw->calibrations[i]); + if (calibration) { + struct tasdevice_data *im = + &(calibration->dev_data); + + if (im->dev_blks) { + struct tasdev_blk *block; + + for (n_blk = 0; n_blk < im->n_blk; + n_blk++) { + block = &(im->dev_blks[n_blk]); + kfree(block->data); + } + kfree(im->dev_blks); + } + } + } + kfree(tas_fmw->calibrations); + } + kfree(tas_fmw); +} + +static int fw_parse_block_data(struct tasdevice_fw *tas_fmw, + struct tasdev_blk *block, const struct firmware *fmw, int offset) +{ + unsigned char *data = (unsigned char *)fmw->data; + int n; + + if (offset + 4 > fmw->size) { + dev_err(tas_fmw->dev, "%s: mnType error\n", __func__); + offset = -1; + goto out; + } + block->type = SMS_HTONL(data[offset], data[offset + 1], + data[offset + 2], data[offset + 3]); + offset += 4; + + if (tas_fmw->fw_hdr.fixed_hdr.drv_ver >= + PPC_DRIVER_CRCCHK) { + if (offset + 1 > fmw->size) { + dev_err(tas_fmw->dev, "%s: mbPChkSumPresent error\n", + __func__); + offset = -1; + goto out; + } + block->is_pchksum_present = data[offset]; + offset++; + + if (offset + 1 > fmw->size) { + dev_err(tas_fmw->dev, "%s: mnPChkSum error\n", + __func__); + offset = -1; + goto out; + } + block->pchksum = data[offset]; + offset++; + + if (offset + 1 > fmw->size) { + dev_err(tas_fmw->dev, "%s: mbYChkSumPresent error\n", + __func__); + offset = -1; + goto out; + } + block->is_ychksum_present = data[offset]; + offset++; + + if (offset + 1 > fmw->size) { + dev_err(tas_fmw->dev, "%s: mnYChkSum error\n", + __func__); + offset = -1; + goto out; + } + block->ychksum = data[offset]; + offset++; + } else { + block->is_pchksum_present = 0; + block->is_ychksum_present = 0; + } + if (offset + 4 > fmw->size) { + dev_err(tas_fmw->dev, "%s: mnCommands error\n", __func__); + offset = -1; + goto out; + } + block->n_cmds = SMS_HTONL(data[offset], data[offset + 1], + data[offset + 2], data[offset + 3]); + offset += 4; + + n = block->n_cmds * 4; + if (offset + n > fmw->size) { + dev_err(tas_fmw->dev, + "%s: File Size(%lu) error offset = %d n = %d\n", + __func__, (unsigned long)fmw->size, offset, n); + offset = -1; + goto out; + } + block->data = kmemdup(&data[offset], n, GFP_KERNEL); + if (block->data == NULL) { + offset = -1; + goto out; + } + offset += n; +out: + return offset; +} + +static int fw_parse_data(struct tasdevice_fw *tas_fmw, + struct tasdevice_data *img_data, const struct firmware *fmw, + int offset) +{ + const unsigned char *data = (unsigned char *)fmw->data; + int n = 0; + unsigned int n_block; + + if (offset + 64 > fmw->size) { + dev_err(tas_fmw->dev, "%s: mpName error\n", __func__); + n = -1; + goto out; + } + memcpy(img_data->name, &data[offset], 64); + offset += 64; + + n = strlen((char *)&data[offset]); + n++; + if (offset + n > fmw->size) { + dev_err(tas_fmw->dev, "%s: mpDescription error\n", __func__); + offset = -1; + goto out; + } + offset += n; + + if (offset + 2 > fmw->size) { + dev_err(tas_fmw->dev, "%s: mnBlocks error\n", __func__); + offset = -1; + goto out; + } + img_data->n_blk = SMS_HTONS(data[offset], data[offset + 1]); + offset += 2; + + img_data->dev_blks = + kcalloc(img_data->n_blk, sizeof(struct tasdev_blk), + GFP_KERNEL); + if (img_data->dev_blks == NULL) { + dev_err(tas_fmw->dev, "%s: FW memory failed!\n", __func__); + goto out; + } + for (n_block = 0; n_block < img_data->n_blk; n_block++) { + offset = fw_parse_block_data(tas_fmw, + &(img_data->dev_blks[n_block]), fmw, offset); + if (offset < 0) { + offset = -1; + goto out; + } + } +out: + return offset; +} + +static int fw_parse_calibration_data(struct tasdevice_priv *tas_dev, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + unsigned char *data = (unsigned char *)fmw->data; + unsigned int n = 0; + unsigned int n_calibration = 0; + struct tasdevice_calibration *calibration = NULL; + + if (offset + 2 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnCalibrations error\n", __func__); + offset = -1; + goto out; + } + tas_fmw->n_calibrations = SMS_HTONS(data[offset], data[offset + 1]); + offset += 2; + + if (tas_fmw->n_calibrations != 1) { + dev_err(tas_dev->dev, + "%s: only support one calibraiton(%d)!\n", + __func__, tas_fmw->n_calibrations); + goto out; + } + + tas_fmw->calibrations = + kcalloc(tas_fmw->n_calibrations, + sizeof(struct tasdevice_calibration), GFP_KERNEL); + if (tas_fmw->calibrations == NULL) { + dev_err(tas_dev->dev, "%s: mpCalibrations memory failed!\n", + __func__); + offset = -1; + goto out; + } + for (n_calibration = 0; n_calibration < tas_fmw->n_calibrations; + n_calibration++) { + if (offset + 64 > fmw->size) { + dev_err(tas_dev->dev, "%s: mpCalibrations error\n", + __func__); + offset = -1; + goto out; + } + calibration = &(tas_fmw->calibrations[n_calibration]); + offset += 64; + + n = strlen((char *)&data[offset]); + n++; + if (offset + n > fmw->size) { + dev_err(tas_dev->dev, "%s: mpDescription error\n", + __func__); + offset = -1; + goto out; + } + offset += n; + + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, + "%s: mnProgram error, offset = %d\n", __func__, + offset); + offset = -1; + goto out; + } + offset++; + + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, + "%s: mnConfiguration error, offset = %d\n", + __func__, + offset); + offset = -1; + goto out; + } + offset++; + + offset = fw_parse_data(tas_fmw, &(calibration->dev_data), fmw, + offset); + if (offset < 0) + goto out; + } + +out: + + return offset; +} + +static int fw_parse_program_data(struct tasdevice_priv *tas_dev, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + struct tasdevice_prog *program; + unsigned char *buf = (unsigned char *)fmw->data; + int n_program = 0; + + if (offset + 2 > fmw->size) { + dev_err(tas_dev->dev, "%s: File Size error\n", __func__); + offset = -1; + goto out; + } + tas_fmw->n_programs = SMS_HTONS(buf[offset], buf[offset + 1]); + offset += 2; + + if (tas_fmw->n_programs == 0) { + dev_err(tas_dev->dev, "%s: mnPrograms is null, maybe calbin\n", + __func__); + //Do not "offset = -1;", because of calbin + goto out; + } + + tas_fmw->programs = + kcalloc(tas_fmw->n_programs, sizeof(struct tasdevice_prog), + GFP_KERNEL); + if (tas_fmw->programs == NULL) { + offset = -1; + goto out; + } + for (n_program = 0; n_program < tas_fmw->n_programs; n_program++) { + int n = 0; + + program = &(tas_fmw->programs[n_program]); + if (offset + 64 > fmw->size) { + dev_err(tas_dev->dev, "%s: mpName error\n", __func__); + offset = -1; + goto out; + } + offset += 64; + + n = strlen((char *)&buf[offset]); + n++; + if (offset + n > fmw->size) { + dev_err(tas_dev->dev, "%s: mpDescription error\n", + __func__); + offset = -1; + goto out; + } + + offset += n; + + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnAppMode error\n", + __func__); + offset = -1; + goto out; + } + offset++; + + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnPDMI2SMode error\n", + __func__); + offset = -1; + goto out; + } + offset++; + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnISnsPD error\n", + __func__); + offset = -1; + goto out; + } + offset++; + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnVSnsPD error\n", + __func__); + offset = -1; + goto out; + } + offset++; + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnPowerLDG error\n", + __func__); + offset = -1; + goto out; + } + offset++; + + offset = fw_parse_data(tas_fmw, &(program->dev_data), fmw, + offset); + if (offset < 0) + goto out; + } +out: + return offset; +} + +static int fw_parse_configuration_data( + struct tasdevice_priv *tas_dev, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset) +{ + unsigned char *data = (unsigned char *)fmw->data; + int n; + unsigned int n_configs; + struct tasdevice_config *config; + + if (offset + 2 > fmw->size) { + dev_err(tas_dev->dev, "%s: File Size error\n", __func__); + offset = -1; + goto out; + } + tas_fmw->n_configurations = SMS_HTONS(data[offset], data[offset + 1]); + offset += 2; + + if (tas_fmw->n_configurations == 0) { + dev_err(tas_dev->dev, "%s: mnConfigurations is zero\n", + __func__); + //Do not "offset = -1;", because of calbin + goto out; + } + tas_fmw->configs = + kcalloc(tas_fmw->n_configurations, + sizeof(struct tasdevice_config), GFP_KERNEL); + + for (n_configs = 0; n_configs < tas_fmw->n_configurations; + n_configs++) { + config = + &(tas_fmw->configs[n_configs]); + if (offset + 64 > fmw->size) { + dev_err(tas_dev->dev, "%s: File Size error\n", + __func__); + offset = -1; + goto out; + } + memcpy(config->name, &data[offset], 64); + offset += 64; + + n = strlen((char *)&data[offset]); + n++; + if (offset + n > fmw->size) { + dev_err(tas_dev->dev, "%s: mpDescription error\n", + __func__); + offset = -1; + goto out; + } + + offset += n; + if (offset + 2 > fmw->size) { + dev_err(tas_dev->dev, + "%s: mnDevice_orientation error\n", + __func__); + offset = -1; + goto out; + } + offset += 2; + + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, "%s: mProgram error\n", + __func__); + offset = -1; + goto out; + } + offset++; + + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnSamplingRate error\n", + __func__); + offset = -1; + goto out; + } + offset += 4; + + if (offset + 1 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnPLLSrc error\n", + __func__); + offset = -1; + goto out; + } + offset++; + + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnPLLSrcRate error\n", + __func__); + offset = -1; + goto out; + } + offset += 4; + + if (offset + 2 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnFsRate error\n", + __func__); + offset = -1; + goto out; + } + offset += 2; + + offset = fw_parse_data(tas_fmw, &(config->dev_data), + fmw, offset); + if (offset < 0) + goto out; + } +out: + return offset; +} + +static int fw_parse_header(struct tasdevice_priv *tas_dev, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + struct tasdevice_dspfw_hdr *fw_hdr = &(tas_fmw->fw_hdr); + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr = &(fw_hdr->fixed_hdr); + const unsigned char *buf = (unsigned char *)fmw->data; + const unsigned char magic_number[] = { 0x35, 0x35, 0x35, 0x32 }; + + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: File Size error\n", __func__); + offset = -EINVAL; + goto out; + } + if (memcmp(&buf[offset], magic_number, 4)) { + dev_err(tas_dev->dev, "%s: Magic number doesn't match", + __func__); + offset = -EINVAL; + goto out; + } + offset += 4; + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnFWSize error\n", __func__); + offset = -1; + goto out; + } + fw_fixed_hdr->fwsize = SMS_HTONL(buf[offset], + buf[offset + 1], buf[offset + 2], buf[offset + 3]); + offset += 4; + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: File Size error\n", __func__); + offset = -1; + goto out; + } + if (fw_fixed_hdr->fwsize != fmw->size) { + dev_err(tas_dev->dev, "File size not match, %lu %u", + (unsigned long)fmw->size, fw_fixed_hdr->fwsize); + offset = -1; + goto out; + } + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnChecksum error\n", __func__); + offset = -1; + goto out; + } + offset += 4; + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnPPCVersion error\n", __func__); + offset = -1; + goto out; + } + fw_fixed_hdr->ppcver = SMS_HTONL(buf[offset], + buf[offset + 1], buf[offset + 2], buf[offset + 3]); + offset += 4; + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnFWVersion error\n", __func__); + offset = -1; + goto out; + } + offset += 4; + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnDriverVersion error\n", __func__); + offset = -1; + goto out; + } + fw_fixed_hdr->drv_ver = SMS_HTONL(buf[offset], + buf[offset + 1], buf[offset + 2], buf[offset + 3]); + offset += 4; + if (offset + 4 > fmw->size) { + dev_err(tas_dev->dev, "%s: mnTimeStamp error\n", __func__); + offset = -1; + goto out; + } + offset += 4; + if (offset + 64 > fmw->size) { + dev_err(tas_dev->dev, "%s: mpDDCName error\n", __func__); + offset = -1; + goto out; + } + offset += 64; + + out: + return offset; +} + +static const unsigned char crc8_lookup_table[CRC8_TABLE_SIZE] = { + 0x00, 0x4D, 0x9A, 0xD7, 0x79, 0x34, 0xE3, 0xAE, + 0xF2, 0xBF, 0x68, 0x25, 0x8B, 0xC6, 0x11, 0x5C, + 0xA9, 0xE4, 0x33, 0x7E, 0xD0, 0x9D, 0x4A, 0x07, + 0x5B, 0x16, 0xC1, 0x8C, 0x22, 0x6F, 0xB8, 0xF5, + 0x1F, 0x52, 0x85, 0xC8, 0x66, 0x2B, 0xFC, 0xB1, + 0xED, 0xA0, 0x77, 0x3A, 0x94, 0xD9, 0x0E, 0x43, + 0xB6, 0xFB, 0x2C, 0x61, 0xCF, 0x82, 0x55, 0x18, + 0x44, 0x09, 0xDE, 0x93, 0x3D, 0x70, 0xA7, 0xEA, + 0x3E, 0x73, 0xA4, 0xE9, 0x47, 0x0A, 0xDD, 0x90, + 0xCC, 0x81, 0x56, 0x1B, 0xB5, 0xF8, 0x2F, 0x62, + 0x97, 0xDA, 0x0D, 0x40, 0xEE, 0xA3, 0x74, 0x39, + 0x65, 0x28, 0xFF, 0xB2, 0x1C, 0x51, 0x86, 0xCB, + 0x21, 0x6C, 0xBB, 0xF6, 0x58, 0x15, 0xC2, 0x8F, + 0xD3, 0x9E, 0x49, 0x04, 0xAA, 0xE7, 0x30, 0x7D, + 0x88, 0xC5, 0x12, 0x5F, 0xF1, 0xBC, 0x6B, 0x26, + 0x7A, 0x37, 0xE0, 0xAD, 0x03, 0x4E, 0x99, 0xD4, + 0x7C, 0x31, 0xE6, 0xAB, 0x05, 0x48, 0x9F, 0xD2, + 0x8E, 0xC3, 0x14, 0x59, 0xF7, 0xBA, 0x6D, 0x20, + 0xD5, 0x98, 0x4F, 0x02, 0xAC, 0xE1, 0x36, 0x7B, + 0x27, 0x6A, 0xBD, 0xF0, 0x5E, 0x13, 0xC4, 0x89, + 0x63, 0x2E, 0xF9, 0xB4, 0x1A, 0x57, 0x80, 0xCD, + 0x91, 0xDC, 0x0B, 0x46, 0xE8, 0xA5, 0x72, 0x3F, + 0xCA, 0x87, 0x50, 0x1D, 0xB3, 0xFE, 0x29, 0x64, + 0x38, 0x75, 0xA2, 0xEF, 0x41, 0x0C, 0xDB, 0x96, + 0x42, 0x0F, 0xD8, 0x95, 0x3B, 0x76, 0xA1, 0xEC, + 0xB0, 0xFD, 0x2A, 0x67, 0xC9, 0x84, 0x53, 0x1E, + 0xEB, 0xA6, 0x71, 0x3C, 0x92, 0xDF, 0x08, 0x45, + 0x19, 0x54, 0x83, 0xCE, 0x60, 0x2D, 0xFA, 0xB7, + 0x5D, 0x10, 0xC7, 0x8A, 0x24, 0x69, 0xBE, 0xF3, + 0xAF, 0xE2, 0x35, 0x78, 0xD6, 0x9B, 0x4C, 0x01, + 0xF4, 0xB9, 0x6E, 0x23, 0x8D, 0xC0, 0x17, 0x5A, + 0x06, 0x4B, 0x9C, 0xD1, 0x7F, 0x32, 0xE5, 0xA8 +}; + +static int check_inpage_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + int ret = 0; + + if (book == TAS2781_YRAM_BOOK1) { + if (page == TAS2781_YRAM1_PAGE) { + if (reg >= TAS2781_YRAM1_START_REG) { + cd->offset = reg; + cd->len = len; + ret = 1; + } else if ((reg + len) > TAS2781_YRAM1_START_REG) { + cd->offset = TAS2781_YRAM1_START_REG; + cd->len = + len - (TAS2781_YRAM1_START_REG - reg); + ret = 1; + } else + ret = 0; + } else if (page == TAS2781_YRAM3_PAGE) { + if (reg > TAS2781_YRAM3_END_REG) { + ret = 0; + } else if (reg >= TAS2781_YRAM3_START_REG) { + if ((reg + len) > TAS2781_YRAM3_END_REG) { + cd->offset = reg; + cd->len = + TAS2781_YRAM3_END_REG - reg + 1; + ret = 1; + } else { + cd->offset = reg; + cd->len = len; + ret = 1; + } + } else { + if ((reg + (len-1)) < + TAS2781_YRAM3_START_REG) + ret = 0; + else { + cd->offset = + TAS2781_YRAM3_START_REG; + cd->len = + len - (TAS2781_YRAM3_START_REG - reg); + ret = 1; + } + } + } + } else if (book == + TAS2781_YRAM_BOOK2) { + if (page == TAS2781_YRAM5_PAGE) { + if (reg > TAS2781_YRAM5_END_REG) { + ret = 0; + } else if (reg >= TAS2781_YRAM5_START_REG) { + if ((reg + len) > TAS2781_YRAM5_END_REG) { + cd->offset = reg; + cd->len = + TAS2781_YRAM5_END_REG - reg + 1; + ret = 1; + } else { + cd->offset = reg; + cd->len = len; + ret = 1; + } + } else { + if ((reg + (len-1)) < + TAS2781_YRAM5_START_REG) + ret = 0; + else { + cd->offset = + TAS2781_YRAM5_START_REG; + cd->len = + len - (TAS2781_YRAM5_START_REG - reg); + ret = 1; + } + } + } + } else + ret = 0; + + return ret; +} + +static int check_inblock_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + int ret = 0; + + if (book == TAS2781_YRAM_BOOK1) { + if (page < TAS2781_YRAM2_START_PAGE) + ret = 0; + else if (page <= TAS2781_YRAM2_END_PAGE) { + if (reg > TAS2781_YRAM2_END_REG) + ret = 0; + else if (reg >= TAS2781_YRAM2_START_REG) { + cd->offset = reg; + cd->len = len; + ret = 1; + } else { + if ((reg + (len-1)) < + TAS2781_YRAM2_START_REG) + ret = 0; + else { + cd->offset = + TAS2781_YRAM2_START_REG; + cd->len = + reg + len - TAS2781_YRAM2_START_REG; + ret = 1; + } + } + } else + ret = 0; + } else if (book == + TAS2781_YRAM_BOOK2) { + if (page < TAS2781_YRAM4_START_PAGE) + ret = 0; + else if (page <= TAS2781_YRAM4_END_PAGE) { + if (reg > TAS2781_YRAM2_END_REG) + ret = 0; + else if (reg >= TAS2781_YRAM2_START_REG) { + cd->offset = reg; + cd->len = len; + ret = 1; + } else { + if ((reg + (len-1)) + < TAS2781_YRAM2_START_REG) + ret = 0; + else { + cd->offset = + TAS2781_YRAM2_START_REG; + cd->len = + reg + len - TAS2781_YRAM2_START_REG; + ret = 1; + } + } + } else + ret = 0; + } else + ret = 0; + + return ret; +} + +static int check_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + int ret = 0; + + ret = check_inpage_yram(cd, book, page, reg, len); + if (ret == 0) + ret = check_inblock_yram(cd, book, + page, reg, len); + + return ret; +} + +static int do_singlereg_checksum(struct tasdevice_priv *tasdevice, + enum channel chl, unsigned char book, unsigned char page, + unsigned char reg, unsigned char val) +{ + int ret = 0; + struct tas_crc crc_data; + unsigned int nData1 = 0; + + if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) + && (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) + && (reg >= TASDEVICE_PAGE_REG(TAS2781_SA_COEFF_SWAP_REG)) + && (reg <= (TASDEVICE_PAGE_REG( + TAS2781_SA_COEFF_SWAP_REG) + 4))) { + /*DSP swap command, pass */ + ret = 0; + goto end; + } + + ret = check_yram(&crc_data, book, page, reg, 1); + if (ret == 1) { + ret = tasdevice_dev_read(tasdevice, chl, + TASDEVICE_REG(book, page, reg), &nData1); + if (ret < 0) + goto end; + + if (nData1 != val) { + dev_err(tasdevice->dev, + "B[0x%x]P[0x%x]R[0x%x] W[0x%x], R[0x%x]\n", + book, page, reg, + val, nData1); + ret = -EAGAIN; + tasdevice->tasdevice[chl].err_code |= + ERROR_YRAM_CRCCHK; + ret = -EAGAIN; + goto end; + } + + ret = crc8(crc8_lookup_table, &val, 1, 0); + } + +end: + return ret; +} + +static int do_multireg_checksum(struct tasdevice_priv *tasdevice, + enum channel chn, unsigned char book, unsigned char page, + unsigned char reg, unsigned int len) +{ + int ret = 0, i = 0; + unsigned char crc_chksum = 0; + unsigned char nBuf1[128] = {0}; + struct tas_crc crc_data; + + if ((reg + len-1) > 127) { + ret = -EINVAL; + dev_err(tasdevice->dev, "firmware error\n"); + goto end; + } + + if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) + && (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) + && (reg == TASDEVICE_PAGE_REG(TAS2781_SA_COEFF_SWAP_REG)) + && (len == 4)) { + /*DSP swap command, pass */ + ret = 0; + goto end; + } + + ret = check_yram(&crc_data, book, page, reg, len); + dev_info(tasdevice->dev, + "isYRAM: Bk 0x%02x, Pg 0x%02x, Rg 0x%02x\n", + book, page, reg); + dev_info(tasdevice->dev, + "isYRAM: crc_data.mnLen 0x%x, len 0x%x, nResult %d\n", + crc_data.len, len, ret); + dev_info(tasdevice->dev, "crc_data.mnOffset %x\n", crc_data.offset); + if (ret == 1) { + if (len == 1) { + dev_err(tasdevice->dev, "firmware error\n"); + ret = -EINVAL; + goto end; + } else { + ret = tasdevice_dev_bulk_read(tasdevice, chn, + TASDEVICE_REG(book, page, crc_data.offset), + nBuf1, crc_data.len); + if (ret < 0) + goto end; + + for (i = 0; i < crc_data.len; i++) { + if ((book == TASDEVICE_BOOK_ID( + TAS2781_SA_COEFF_SWAP_REG)) + && (page == TASDEVICE_PAGE_ID( + TAS2781_SA_COEFF_SWAP_REG)) + && ((i + crc_data.offset) + >= TASDEVICE_PAGE_REG( + TAS2781_SA_COEFF_SWAP_REG)) + && ((i + crc_data.offset) + <= (TASDEVICE_PAGE_REG( + TAS2781_SA_COEFF_SWAP_REG) + + 4))) { + /*DSP swap command, bypass */ + continue; + } else + crc_chksum += + crc8(crc8_lookup_table, + &nBuf1[i], 1, 0); + } + + ret = crc_chksum; + } + } + +end: + return ret; +} + +static int tasdevice_load_block(struct tasdevice_priv *tas_dev, + struct tasdev_blk *block) +{ + int ret = 0; + unsigned int n_cmds = 0; + unsigned char book = 0; + unsigned char page = 0; + unsigned char offset = 0; + unsigned char val = 0; + unsigned int len = 0; + unsigned int sleep_dur = 0; + unsigned char crc_chksum = 0; + unsigned int n_value = 0; + int n_retry = 6; + unsigned char *data = block->data; + int chn = 0, chnend = 0; + + dev_info(tas_dev->dev, + "load block: Type = %d, commands = %d\n", + block->type, block->n_cmds); + switch (block->type) { + case MAIN_ALL_DEVICES: + chn = 0; + chnend = tas_dev->ndev; + break; + case MAIN_DEVICE_A: + case COEFF_DEVICE_A: + case PRE_DEVICE_A: + chn = 0; + chnend = 1; + break; + case MAIN_DEVICE_B: + case COEFF_DEVICE_B: + case PRE_DEVICE_B: + chn = 1; + chnend = 2; + break; + case MAIN_DEVICE_C: + case COEFF_DEVICE_C: + case PRE_DEVICE_C: + chn = 2; + chnend = 3; + break; + case MAIN_DEVICE_D: + case COEFF_DEVICE_D: + case PRE_DEVICE_D: + chn = 3; + chnend = 4; + break; + default: + dev_info(tas_dev->dev, + "load block: Other Type = 0x%02x\n", + block->type); + break; + } + + for (; chn < chnend; chn++) { + if (tas_dev->tasdevice[chn].is_loading == false) + continue; +start: + if (block->is_pchksum_present) { + ret = tasdevice_dev_write(tas_dev, chn, + TASDEVICE_I2CChecksum, 0); + if (ret < 0) + goto end; + } + + if (block->is_ychksum_present) + crc_chksum = 0; + + n_cmds = 0; + + while (n_cmds < block->n_cmds) { + data = block->data + n_cmds * 4; + + book = data[0]; + page = data[1]; + offset = data[2]; + val = data[3]; + + n_cmds++; + + if (offset <= 0x7F) { + ret = tasdevice_dev_write(tas_dev, chn, + TASDEVICE_REG(book, page, offset), + val); + if (ret < 0) + goto end; + if (block->is_ychksum_present) { + ret = do_singlereg_checksum(tas_dev, + chn, book, page, offset, + val); + if (ret < 0) + goto check; + crc_chksum += (unsigned char)ret; + } + } else if (offset == 0x81) { + sleep_dur = (book << 8) + page; + msleep(sleep_dur); + } else if (offset == 0x85) { + data += 4; + len = (book << 8) + page; + book = data[0]; + page = data[1]; + offset = data[2]; + if (len > 1) { + ret = tasdevice_dev_bulk_write( + tas_dev, chn, + TASDEVICE_REG(book, page, + offset), data + 3, len); + if (ret < 0) + goto end; + if (block->is_ychksum_present) + ret = do_multireg_checksum( + tas_dev, chn, book, + page, offset, + len); + } else { + ret = tasdevice_dev_write(tas_dev, + chn, + TASDEVICE_REG(book, page, + offset), + data[3]); + if (ret < 0) + goto end; + if (block->is_ychksum_present) + ret = do_singlereg_checksum( + tas_dev, chn, book, + page, offset, + data[3]); + } + if (block->is_ychksum_present) { + if (ret < 0) + goto check; + crc_chksum += (unsigned char)ret; + } + + n_cmds++; + if (len >= 2) + n_cmds += ((len - 2) / 4) + 1; + } + } + if (block->is_pchksum_present) { + ret = tasdevice_dev_read(tas_dev, chn, + TASDEVICE_I2CChecksum, &n_value); + if (ret < 0) { + dev_err(tas_dev->dev, "%s: Channel %d\n", + __func__, chn); + goto check; + } + if ((n_value & 0xff) != block->pchksum) { + dev_err(tas_dev->dev, + "Block PChkSum Channel %d ", chn); + dev_err(tas_dev->dev, + "FW = 0x%x, Reg = 0x%x\n", + block->pchksum, + (n_value & 0xff)); + ret = -EAGAIN; + tas_dev->tasdevice[chn].err_code |= + ERROR_PRAM_CRCCHK; + goto check; + } + ret = 0; + tas_dev->tasdevice[chn].err_code &= + ~ERROR_PRAM_CRCCHK; + dev_info(tas_dev->dev, "Block[0x%02x] PChkSum match\n", + block->type); + } + + if (block->is_ychksum_present) { + //TBD, open it when FW ready + dev_err(tas_dev->dev, + "Block YChkSum: FW = 0x%x, YCRC = 0x%x\n", + block->ychksum, crc_chksum); + + tas_dev->tasdevice[chn].err_code &= + ~ERROR_YRAM_CRCCHK; + ret = 0; + dev_info(tas_dev->dev, + "Block[0x%x] YChkSum match\n", block->type); + } +check: + if (ret == -EAGAIN) { + n_retry--; + if (n_retry > 0) + goto start; + else { + if ((block->type == MAIN_ALL_DEVICES) + || (block->type == MAIN_DEVICE_A) + || (block->type == MAIN_DEVICE_B) + || (block->type == MAIN_DEVICE_C) + || (block->type == MAIN_DEVICE_D)) + tas_dev->tasdevice[chn].cur_prog = -1; + else + tas_dev->tasdevice[chn].cur_conf = -1; + n_retry = 6; + } + } + } +end: + if (ret < 0) + dev_err(tas_dev->dev, "Block (%d) load error\n", block->type); + + return ret; +} + + +static int tasdevice_load_data(struct tasdevice_priv *tas_dev, + struct tasdevice_data *dev_data) +{ + int ret = 0; + unsigned int n_block = 0; + struct tasdev_blk *block = NULL; + + dev_info(tas_dev->dev, "%s: TAS2781 load data: %s, Blocks = %d\n", + __func__, dev_data->name, dev_data->n_blk); + + for (n_block = 0; n_block < dev_data->n_blk; n_block++) { + block = &(dev_data->dev_blks[n_block]); + ret = tas_dev->tasdevice_load_block(tas_dev, block); + if (ret < 0) + break; + } + + return ret; +} + +static int tasdevice_load_calibrated_data( + struct tasdevice_priv *tas_dev, struct tasdevice_data *dev_data) +{ + int ret = 0; + unsigned int n_block = 0; + struct tasdev_blk *block = NULL; + + dev_info(tas_dev->dev, "%s: TAS2781 load data: %s, Blocks = %d\n", + __func__, dev_data->name, dev_data->n_blk); + + for (n_block = 0; n_block < dev_data->n_blk; n_block++) { + block = &(dev_data->dev_blks[n_block]); + ret = tasdevice_load_block(tas_dev, block); + if (ret < 0) + break; + } + + return ret; +} + +int tas2781_load_calibration(void *context, + char *file_name, enum channel i) +{ + int ret = 0, offset = 0; + struct firmware fmw; + const struct firmware *fw_entry = NULL; + struct tasdevice_priv *tas_dev = (struct tasdevice_priv *)context; + struct tasdevice *tasdev = &(tas_dev->tasdevice[i]); + struct tasdevice_fw *tas_fmw = NULL; + + ret = request_firmware(&fw_entry, file_name, tas_dev->dev); + if (!ret) { + if (!fw_entry->size) { + dev_err(tas_dev->dev, + "%s: file read error: size = %lu\n", + __func__, (unsigned long)fw_entry->size); + goto out; + } + fmw.size = fw_entry->size; + fmw.data = fw_entry->data; + dev_info(tas_dev->dev, + "%s: file = %s, file size %zd\n", + __func__, file_name, fw_entry->size); + } else { + dev_info(tas_dev->dev, + "%s: Request firmware %s failed\n", + __func__, file_name); + goto out; + } + + tas_fmw = tasdev->cali_data_fmw = kcalloc(1, + sizeof(struct tasdevice_fw), GFP_KERNEL); + if (tasdev->cali_data_fmw == NULL) { + dev_err(tas_dev->dev, "%s: FW memory failed!\n", __func__); + ret = -1; + goto out; + } + tas_fmw->dev = tas_dev->dev; + offset = fw_parse_header(tas_dev, tas_fmw, &fmw, offset); + if (offset == -1) { + dev_err(tas_dev->dev, "%s: fw_parse_header EXIT!\n", __func__); + goto out; + } + offset = fw_parse_variable_hdr_cal(tas_dev, tas_fmw, &fmw, offset); + if (offset == -1) { + dev_err(tas_dev->dev, + "%s: fw_parse_variable_header_cal EXIT!\n", __func__); + goto out; + } + offset = fw_parse_program_data(tas_dev, tas_fmw, &fmw, offset); + if (offset == -1) { + dev_err(tas_dev->dev, "%s: fw_parse_program_data EXIT!\n", + __func__); + goto out; + } + offset = fw_parse_configuration_data(tas_dev, tas_fmw, &fmw, + offset); + if (offset == -1) { + dev_err(tas_dev->dev, + "%s: fw_parse_configuration_data EXIT!\n", __func__); + goto out; + } + offset = fw_parse_calibration_data(tas_dev, + tas_fmw, &fmw, offset); + if (offset == -1) { + dev_err(tas_dev->dev, "%s: fw_parse_calibration_data EXIT!\n", + __func__); + goto out; + } + tasdev->is_calibrated_data_loaded = true; +out: + if (fw_entry) { + release_firmware(fw_entry); + fw_entry = NULL; + } + return ret; +} + +int tasdevice_dspfw_ready(const struct firmware *fmw, void *context) +{ + struct tasdevice_priv *tas_dev = (struct tasdevice_priv *) context; + struct tasdevice_fw *tas_fmw = NULL; + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr = NULL; + int offset = 0, ret = 0; + + if (!fmw || !fmw->data) { + dev_err(tas_dev->dev, "%s: Failed to read firmware %s\n", + __func__, tas_dev->coef_binaryname); + ret = -1; + goto out; + } + + tas_dev->fmw = kcalloc(1, + sizeof(struct tasdevice_fw), GFP_KERNEL); + if (tas_dev->fmw == NULL) { + ret = -1; + goto out; + } + tas_fmw = tas_dev->fmw; + tas_fmw->dev = tas_dev->dev; + offset = fw_parse_header(tas_dev, tas_fmw, fmw, offset); + + if (offset == -1) + goto out; + fw_fixed_hdr = &(tas_fmw->fw_hdr.fixed_hdr); + switch (fw_fixed_hdr->drv_ver) { + case 0x301: + case 0x302: + case 0x502: + tas_dev->fw_parse_variable_header = + fw_parse_variable_header_kernel; + tas_dev->fw_parse_program_data = + fw_parse_program_data_kernel; + tas_dev->fw_parse_configuration_data = + fw_parse_configuration_data_kernel; + tas_dev->tasdevice_load_block = + tasdevice_load_block_kernel; + break; + case 0x202: + case 0x400: + tas_dev->fw_parse_variable_header = + fw_parse_variable_header_git; + tas_dev->fw_parse_program_data = + fw_parse_program_data; + tas_dev->fw_parse_configuration_data = + fw_parse_configuration_data; + tas_dev->tasdevice_load_block = + tasdevice_load_block; + break; + default: + if (fw_fixed_hdr->drv_ver == 0x100) { + if (fw_fixed_hdr->ppcver >= PPC3_VERSION) { + tas_dev->fw_parse_variable_header = + fw_parse_variable_header_kernel; + tas_dev->fw_parse_program_data = + fw_parse_program_data_kernel; + tas_dev->fw_parse_configuration_data = + fw_parse_configuration_data_kernel; + tas_dev->tasdevice_load_block = + tasdevice_load_block_kernel; + tas_dev->fw_parse_calibration_data = NULL; + } else { + switch (fw_fixed_hdr->ppcver) { + case 0x00: + tas_dev->fw_parse_variable_header = + fw_parse_variable_header_git; + tas_dev->fw_parse_program_data = + fw_parse_program_data; + tas_dev->fw_parse_configuration_data = + fw_parse_configuration_data; + tas_dev->fw_parse_calibration_data = + fw_parse_calibration_data; + tas_dev->tasdevice_load_block = + tasdevice_load_block; + break; + default: + dev_err(tas_dev->dev, + "%s: PPCVersion must be 0x0 or 0x%02x", + __func__, PPC3_VERSION); + dev_err(tas_dev->dev, " Current:0x%02x\n", + fw_fixed_hdr->ppcver); + offset = -1; + break; + } + } + } else { + dev_err(tas_dev->dev, + "%s: DriverVersion must be 0x0, 0x230 or above 0x230 ", + __func__); + dev_err(tas_dev->dev, "current is 0x%02x\n", + fw_fixed_hdr->drv_ver); + offset = -1; + } + break; + } + + offset = tas_dev->fw_parse_variable_header(tas_dev, fmw, offset); + if (offset == -1) + goto out; + + offset = tas_dev->fw_parse_program_data(tas_dev, tas_fmw, fmw, + offset); + if (offset < 0) { + ret = -1; + goto out; + } + offset = tas_dev->fw_parse_configuration_data(tas_dev, + tas_fmw, fmw, offset); + if (offset < 0) + ret = -1; + +out: + return ret; +} + +void tasdevice_calbin_remove(void *context) +{ + struct tasdevice_priv *tas_dev = (struct tasdevice_priv *) context; + struct tasdevice *tasdev = NULL; + int i = 0; + + if (tas_dev) { + for (i = 0; i < tas_dev->ndev; i++) { + tasdev = &(tas_dev->tasdevice[i]); + if (tasdev->cali_data_fmw) { + tas2781_clear_calfirmware( + tasdev->cali_data_fmw); + tasdev->cali_data_fmw = NULL; + } + } + } +} + +void tasdevice_dsp_remove(void *context) +{ + struct tasdevice_priv *tas_dev = (struct tasdevice_priv *) context; + int i = 0; + + if (tas_dev->fmw) { + struct tasdevice_fw *tas_fmw = tas_dev->fmw; + + if (tas_fmw->programs) { + struct tasdevice_prog *program; + + for (i = 0; i < tas_fmw->n_programs; i++) { + program = &(tas_fmw->programs[i]); + if (program) { + struct tasdevice_data + *im = &(program->dev_data); + struct tasdev_blk *blk; + unsigned int n_blk; + + if (!im->dev_blks) + continue; + + for (n_blk = 0; n_blk < im->n_blk; + n_blk++) { + blk = &(im->dev_blks[n_blk]); + kfree(blk->data); + } + kfree(im->dev_blks); + } + } + kfree(tas_fmw->programs); + } + + if (tas_fmw->configs) { + struct tasdevice_config *config; + + for (i = 0; i < tas_fmw->n_configurations; i++) { + config = &(tas_fmw->configs[i]); + if (config) { + struct tasdevice_data + *im = &(config->dev_data); + struct tasdev_blk *block; + unsigned int n_blk; + + if (!im->dev_blks) + continue; + + for (n_blk = 0; n_blk < im->n_blk; + n_blk++) { + block = &(im->dev_blks[n_blk]); + kfree(block->data); + } + kfree(im->dev_blks); + } + } + kfree(tas_fmw->configs); + } + kfree(tas_fmw); + tas_dev->fmw = NULL; + } +} + +static int tas2781_set_calibration(void *context, enum channel i, + int n_calibration) +{ + struct tasdevice_priv *tas_dev = (struct tasdevice_priv *) context; + int ret = 0; + struct tasdevice *tasdevice = &(tas_dev->tasdevice[i]); + struct tasdevice_fw *cal_fmw = tasdevice->cali_data_fmw; + + if ((!tas_dev->fmw->programs) + || (!tas_dev->fmw->configs)) { + dev_err(tas_dev->dev, "%s, Firmware not loaded\n\r", __func__); + ret = 0; + goto out; + } + + if (n_calibration == 0xFF || (n_calibration == 0x100 + && tasdevice->is_calibrated_data_loaded == false)) { + if (cal_fmw) { + tasdevice->is_calibrated_data_loaded = false; + tas2781_clear_calfirmware(cal_fmw); + cal_fmw = NULL; + } + + scnprintf(tas_dev->cal_binaryname[i], 64, "%s_cal_0x%02x.bin", + tas_dev->dev_name, tas_dev->tasdevice[i].dev_addr); + ret = tas2781_load_calibration(tas_dev, + tas_dev->cal_binaryname[i], i); + if (ret != 0) { + dev_err(tas_dev->dev, + "%s: load %s error, no-side effect\n", + __func__, tas_dev->cal_binaryname[i]); + ret = 0; + } + } + tasdevice->is_loading = true; + tasdevice->is_loaderr = false; + + if (cal_fmw) { + struct tasdevice_calibration *calibration = + cal_fmw->calibrations; + + if (calibration) + tasdevice_load_calibrated_data(tas_dev, + &(calibration->dev_data)); + } else + dev_err(tas_dev->dev, + "%s: No calibrated data for device %d\n", __func__, i); + +out: + return ret; +} + +int tasdevice_select_tuningprm_cfg(void *context, int prm_no, + int cfg_no, int rca_conf_no) +{ + struct tasdevice_priv *tas_dev = (struct tasdevice_priv *) context; + struct tasdevice_rca *rca = &(tas_dev->rcabin); + struct tasdevice_config_info **cfg_info = rca->cfg_info; + struct tasdevice_fw *tas_fmw = tas_dev->fmw; + struct tasdevice_config *conf = NULL; + struct tasdevice_prog *program = NULL; + int i = 0; + int status = 0; + int prog_status = 0; + + if (tas_fmw == NULL) { + dev_err(tas_dev->dev, "%s: Firmware is NULL\n", __func__); + goto out; + } + + if (cfg_no >= tas_fmw->n_configurations) { + dev_err(tas_dev->dev, + "%s: cfg(%d) is not in range of conf %u\n", + __func__, cfg_no, tas_fmw->n_configurations); + goto out; + } + + if (prm_no >= tas_fmw->n_programs) { + dev_err(tas_dev->dev, + "%s: prm(%d) is not in range of Programs %u\n", + __func__, prm_no, tas_fmw->n_programs); + goto out; + } + + if (rca_conf_no > rca->ncfgs || rca_conf_no <= 0 || + cfg_info == NULL) { + dev_err(tas_dev->dev, + "conf_no:%d should be in range from 0 to %u\n", + rca_conf_no, rca->ncfgs-1); + goto out; + } else { + dev_info(tas_dev->dev, "%s: rca_profile_conf_id = %d\n", + __func__, rca_conf_no); + } + + conf = &(tas_fmw->configs[cfg_no]); + for (i = 0; i < tas_dev->ndev; i++) { + if (cfg_info[rca_conf_no]->active_dev & (1 << i)) { + if (tas_dev->tasdevice[i].cur_prog != prm_no) { + tas_dev->tasdevice[i].cur_conf = -1; + tas_dev->tasdevice[i].is_loading = true; + prog_status++; + } + } else + tas_dev->tasdevice[i].is_loading = false; + tas_dev->tasdevice[i].is_loaderr = false; + } + + if (prog_status) { + program = &(tas_fmw->programs[prm_no]); + tasdevice_load_data(tas_dev, &(program->dev_data)); + for (i = 0; i < tas_dev->ndev; i++) { + if (tas_dev->tasdevice[i].is_loaderr == true) + continue; + else if (tas_dev->tasdevice[i].is_loaderr == false + && tas_dev->tasdevice[i].is_loading == true) { + struct tasdevice_fw *cal_fmw = + tas_dev->tasdevice[i].cali_data_fmw; + + if (cal_fmw) { + struct tasdevice_calibration + *calibration = + cal_fmw->calibrations; + + if (calibration) + tasdevice_load_calibrated_data( + tas_dev, + &(calibration->dev_data)); + } + tas_dev->tasdevice[i].cur_prog = prm_no; + } + } + } + + if (tas_dev->is_glb_calibrated_data_loaded == false) { + for (i = 0; i < tas_dev->ndev; i++) + tas2781_set_calibration(tas_dev, i, 0x100); + tas_dev->is_glb_calibrated_data_loaded = true; + /* No wise to reload calibrationdata everytime, + * this code will work once even if calibrated + * data still failed to be got + */ + } + + for (i = 0; i < tas_dev->ndev; i++) { + dev_info(tas_dev->dev, "%s,fun %d,%d,%d\n", __func__, + tas_dev->tasdevice[i].cur_conf, + cfg_info[rca_conf_no]->active_dev, + tas_dev->tasdevice[i].is_loaderr); + if (tas_dev->tasdevice[i].cur_conf != cfg_no + && (cfg_info[rca_conf_no]->active_dev & (1 << i)) + && (tas_dev->tasdevice[i].is_loaderr == false)) { + status++; + tas_dev->tasdevice[i].is_loading = true; + } else + tas_dev->tasdevice[i].is_loading = false; + } + + if (status) { + status = 0; + tasdevice_load_data(tas_dev, &(conf->dev_data)); + for (i = 0; i < tas_dev->ndev; i++) { + if (tas_dev->tasdevice[i].is_loaderr == true) { + status |= 1 << (i + 4); + continue; + } else if (tas_dev->tasdevice[i].is_loaderr == false + && tas_dev->tasdevice[i].is_loading == true) + tas_dev->tasdevice[i].cur_conf = cfg_no; + } + } else { + dev_err(tas_dev->dev, + "%s: No device is in active in conf %d\n", + __func__, rca_conf_no); + } + + status |= cfg_info[rca_conf_no]->active_dev; + dev_info(tas_dev->dev, "%s: DSP mode: load status is %08x\n", + __func__, status); +out: + return prog_status; +} diff --git a/sound/soc/codecs/tas2781-dsp.h b/sound/soc/codecs/tas2781-dsp.h new file mode 100644 index 000000000000..639e84e0b5b4 --- /dev/null +++ b/sound/soc/codecs/tas2781-dsp.h @@ -0,0 +1,178 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier + * + * Copyright (C) 2022 - 2023 Texas Instruments Incorporated + * https://www.ti.com + * + * The TAS2781 driver implements a flexible and configurable algo coff setting + * for one, two, even multiple TAS2781 chips. + * + * Author: Shenghao Ding <shenghao-ding(a)ti.com> + * Kevin Lu <kevin-lu(a)ti.com> + */ + +#ifndef __TASDEVICE_DSP_H__ +#define __TASDEVICE_DSP_H__ + +#define MAIN_ALL_DEVICES 0x0d +#define MAIN_DEVICE_A 0x01 +#define MAIN_DEVICE_B 0x08 +#define MAIN_DEVICE_C 0x10 +#define MAIN_DEVICE_D 0x14 +#define COEFF_DEVICE_A 0x03 +#define COEFF_DEVICE_B 0x0a +#define COEFF_DEVICE_C 0x11 +#define COEFF_DEVICE_D 0x15 +#define PRE_DEVICE_A 0x04 +#define PRE_DEVICE_B 0x0b +#define PRE_DEVICE_C 0x12 +#define PRE_DEVICE_D 0x16 + +#define PPC3_VERSION 0x4100 +#define RCA_CONFIGID_BYPASS_ALL 0 +#define TASDEVICE_DEVICE_SUM 8 +#define TASDEVICE_CONFIG_SUM 64 + +enum channel { + top_left_Chn, + top_right_chn, + bottom_left_Chn, + bottom_right_chn, + max_chn, +}; + +enum tasdevice_dsp_dev_idx { + TASDEVICE_DSP_TAS_2555 = 0, + TASDEVICE_DSP_TAS_2555_STEREO, + TASDEVICE_DSP_TAS_2557_MONO, + TASDEVICE_DSP_TAS_2557_DUAL_MONO, + TASDEVICE_DSP_TAS_2559, + TASDEVICE_DSP_TAS_2563, + TASDEVICE_DSP_TAS_2563_DUAL_MONO = 7, + TASDEVICE_DSP_TAS_2563_QUAD, + TASDEVICE_DSP_TAS_2563_21, + TASDEVICE_DSP_TAS_2781, + TASDEVICE_DSP_TAS_2781_DUAL_MONO, + TASDEVICE_DSP_TAS_2781_21, + TASDEVICE_DSP_TAS_2781_QUAD, + TASDEVICE_DSP_TAS_MAX_DEVICE +}; + +struct tasdevice_fw_fixed_hdr { + unsigned int fwsize; + unsigned int ppcver; + unsigned int drv_ver; +}; + +struct tasdevice_dspfw_hdr { + struct tasdevice_fw_fixed_hdr fixed_hdr; + unsigned short device_family; + unsigned short device; + unsigned char ndev; +}; + +struct tasdev_blk { + unsigned int type; + unsigned char is_pchksum_present; + unsigned char pchksum; + unsigned char is_ychksum_present; + unsigned char ychksum; + unsigned int n_cmds; + unsigned int blk_size; + unsigned int n_subblocks; + unsigned char *data; +}; + +struct tasdevice_data { + char name[64]; + unsigned int n_blk; + struct tasdev_blk *dev_blks; +}; +struct tasdevice_prog { + unsigned int prog_size; + struct tasdevice_data dev_data; +}; + +struct tasdevice_config { + unsigned int cfg_size; + char name[64]; + struct tasdevice_data dev_data; +}; + +struct tasdevice_calibration { + struct tasdevice_data dev_data; +}; + +struct tasdevice_fw { + struct tasdevice_dspfw_hdr fw_hdr; + unsigned short n_programs; + struct tasdevice_prog *programs; + unsigned short n_configurations; + struct tasdevice_config *configs; + unsigned short n_calibrations; + struct tasdevice_calibration *calibrations; + struct device *dev; +}; + +enum tasdevice_dsp_fw_state { + TASDEVICE_DSP_FW_NONE = 0, + TASDEVICE_DSP_FW_PENDING, + TASDEVICE_DSP_FW_FAIL, + TASDEVICE_DSP_FW_ALL_OK, +}; + +enum tasdevice_bin_blk_type { + TASDEVICE_BIN_BLK_COEFF = 1, + TASDEVICE_BIN_BLK_POST_POWER_UP, + TASDEVICE_BIN_BLK_PRE_SHUTDOWN, + TASDEVICE_BIN_BLK_PRE_POWER_UP, + TASDEVICE_BIN_BLK_POST_SHUTDOWN +}; + +struct tasdevice_rca_hdr { + unsigned int img_sz; + unsigned int checksum; + unsigned int binary_version_num; + unsigned int drv_fw_version; + unsigned char plat_type; + unsigned char dev_family; + unsigned char reserve; + unsigned char ndev; + unsigned char devs[TASDEVICE_DEVICE_SUM]; + unsigned int nconfig; + unsigned int config_size[TASDEVICE_CONFIG_SUM]; +}; + +struct tasdev_blk_data { + unsigned char dev_idx; + unsigned char block_type; + unsigned short yram_checksum; + unsigned int block_size; + unsigned int n_subblks; + unsigned char *regdata; +}; + +struct tasdevice_config_info { + unsigned int nblocks; + unsigned int real_nblocks; + unsigned char active_dev; + struct tasdev_blk_data **blk_data; +}; + +struct tasdevice_rca { + struct tasdevice_rca_hdr fw_hdr; + int ncfgs; + struct tasdevice_config_info **cfg_info; + int profile_cfg_id; +}; + +int tasdevice_dspfw_ready(const struct firmware *fmw, void *context); +void tasdevice_dsp_remove(void *context); +void tasdevice_calbin_remove(void *context); +int tas2781_load_calibration(void *tas_dev, char *filename, + enum channel i); +int tasdevice_select_tuningprm_cfg(void *context, int prm, + int cfg_no, int rca_conf_no); + +#endif diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c new file mode 100644 index 000000000000..2b2b8367044f --- /dev/null +++ b/sound/soc/codecs/tas2781-i2c.c @@ -0,0 +1,1913 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier + * + * Copyright (C) 2022 - 2023 Texas Instruments Incorporated + * https://www.ti.com + * + * The TAS2781 driver implements a flexible and configurable algo coff setting + * for one, two, even multiple TAS2781 chips. + * + * Author: Shenghao Ding <shenghao-ding(a)ti.com> + * Kevin Lu <kevin-lu(a)ti.com> + */ + +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/init.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/of.h> +#include <linux/slab.h> +#include <linux/firmware.h> +#include <linux/of_gpio.h> +#include <linux/interrupt.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> + +#include "tas2781.h" + +/* max. length of the alsa mixer control name */ +#define MAX_CONTROL_NAME (48) +#define TASDEVICE_CLK_DIR_IN (0) +#define TASDEVICE_CLK_DIR_OUT (1) + +#define TASDEVICE_IRQ_DET_TIMEOUT (30000) +#define TASDEVICE_IRQ_DET_CNT_LIMIT (500) +#define TAS2781_REG_INT_LTCH0 TASDEVICE_REG(0X0, 0x0, 0x49) +#define TAS2781_REG_INT_LTCH1 TASDEVICE_REG(0X0, 0x0, 0x4A) +#define TAS2781_REG_INT_LTCH1_0 TASDEVICE_REG(0X0, 0x0, 0x4B) +#define TAS2781_REG_INT_LTCH2 TASDEVICE_REG(0X0, 0x0, 0x4F) +#define TAS2781_REG_INT_LTCH3 TASDEVICE_REG(0X0, 0x0, 0x50) +#define TAS2781_REG_INT_LTCH4 TASDEVICE_REG(0X0, 0x0, 0x51) + +const char *blocktype[5] = { + "COEFF", + "POST_POWER_UP", + "PRE_SHUTDOWN", + "PRE_POWER_UP", + "POST_SHUTDOWN" +}; + +static const char * const dts_dev_addr_tag[] = { + "topleft-channel", + "topright-channel", + "bottomleft-channel", + "bottomright-channel" +}; + +static const struct i2c_device_id tasdevice_id[] = { + { "tas2781", TAS2781 }, + {} +}; + +MODULE_DEVICE_TABLE(i2c, tasdevice_id); + +static const struct regmap_range_cfg tasdevice_ranges[] = { + { + .range_min = 0, + .range_max = 256 * 128, + .selector_reg = TASDEVICE_PAGE_SELECT, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = 128, + }, +}; + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 8, + .val_bits = 8, + .cache_type = REGCACHE_RBTREE, + .ranges = tasdevice_ranges, + .num_ranges = ARRAY_SIZE(tasdevice_ranges), + .max_register = 256 * 128, +}; + +static const struct of_device_id tasdevice_of_match[] = { + { .compatible = "ti,tas2781" }, + {}, +}; + +MODULE_DEVICE_TABLE(of, tasdevice_of_match); + +int tasdevice_process_block(void *context, + unsigned char *data, unsigned char dev_idx, int sublocksize) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *)context; + unsigned char subblk_typ = data[1]; + int subblk_offset = 2; + int chn = 0, chnend = 0; + int rc = 0; + int blktyp = dev_idx & 0xC0, idx = dev_idx & 0x3F; + bool is_err = false; + + if (idx) { + chn = idx-1; + chnend = idx; + } else { + if (tas_dev->set_global_mode) { + chn = tas_dev->ndev; + chnend = tas_dev->ndev + 1; + } else { + chn = 0; + chnend = tas_dev->ndev; + } + } + + for (; chn < chnend; chn++) { + if (tas_dev->set_global_mode == NULL && + tas_dev->tasdevice[chn].is_loading == false) + continue; + + is_err = false; + subblk_offset = 2; + switch (subblk_typ) { + case TASDEVICE_CMD_SING_W: { + int i = 0; + unsigned short len = SMS_HTONS(data[2], data[3]); + + subblk_offset += 2; + if (subblk_offset + 4 * len > sublocksize) { + dev_err(tas_dev->dev, + "process_block: Out of memory\n"); + is_err = true; + break; + } + + for (i = 0; i < len; i++) { + rc = tasdevice_dev_write(tas_dev, chn, + TASDEVICE_REG(data[subblk_offset], + data[subblk_offset + 1], + data[subblk_offset + 2]), + data[subblk_offset + 3]); + if (rc < 0) { + is_err = true; + dev_err(tas_dev->dev, + "process_block: single write error\n"); + } + subblk_offset += 4; + } + } + break; + case TASDEVICE_CMD_BURST: { + unsigned short len = + SMS_HTONS(data[2], data[3]); + + subblk_offset += 2; + if (subblk_offset + 4 + len > sublocksize) { + dev_err(tas_dev->dev, + "%s: BURST Out of memory\n", __func__); + is_err = true; + break; + } + if (len % 4) { + dev_err(tas_dev->dev, + "%s:Bst-len(%u)not div by 4\n", + __func__, len); + break; + } + + rc = tasdevice_dev_bulk_write(tas_dev, chn, + TASDEVICE_REG(data[subblk_offset], + data[subblk_offset + 1], + data[subblk_offset + 2]), + &(data[subblk_offset + 4]), + len); + if (rc < 0) { + is_err = true; + dev_err(tas_dev->dev, + "%s: bulk_write error = %d\n", + __func__, rc); + } + subblk_offset += (len + 4); + } + break; + case TASDEVICE_CMD_DELAY: { + unsigned short delay_time = 0; + + if (subblk_offset + 2 > sublocksize) { + dev_err(tas_dev->dev, + "%s: deley Out of memory\n", + __func__); + is_err = true; + break; + } + delay_time = SMS_HTONS(data[2], data[3]); + usleep_range(delay_time*1000, delay_time*1000); + subblk_offset += 2; + } + break; + case TASDEVICE_CMD_FIELD_W: + if (subblk_offset + 6 > sublocksize) { + dev_err(tas_dev->dev, + "%s: bit write Out of memory\n", + __func__); + is_err = true; + break; + } + rc = tasdevice_dev_update_bits(tas_dev, chn, + TASDEVICE_REG(data[subblk_offset + 2], + data[subblk_offset + 3], + data[subblk_offset + 4]), + data[subblk_offset + 1], + data[subblk_offset + 5]); + if (rc < 0) { + is_err = true; + dev_err(tas_dev->dev, + "%s: update_bits error = %d\n", + __func__, rc); + } + subblk_offset += 6; + break; + default: + break; + }; + if (is_err == true && blktyp != 0) { + if (blktyp == 0x80) { + tas_dev->tasdevice[chn].cur_prog = -1; + tas_dev->tasdevice[chn].cur_conf = -1; + } else + tas_dev->tasdevice[chn].cur_conf = -1; + } + } + return subblk_offset; +} + +static void tasdevice_select_cfg_blk(void *pContext, int conf_no, + unsigned char block_type) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) pContext; + struct tasdevice_rca *rca = &(tas_dev->rcabin); + struct tasdevice_config_info **cfg_info = rca->cfg_info; + int j = 0, k = 0, chn = 0, chnend = 0; + + if (conf_no >= rca->ncfgs || conf_no < 0 || NULL == cfg_info) { + dev_err(tas_dev->dev, + "conf_no should be not more than %u\n", + rca->ncfgs); + goto out; + } else + dev_info(tas_dev->dev, + "select_cfg_blk: profile_conf_id = %d\n", + conf_no); + + for (j = 0; j < (int)cfg_info[conf_no]->real_nblocks; j++) { + unsigned int length = 0, rc = 0; + + if (block_type > 5 || block_type < 2) { + dev_err(tas_dev->dev, + "ERROR!!!block_type should be in range from 2 to 5\n"); + goto out; + } + if (block_type != cfg_info[conf_no]->blk_data[j]->block_type) + continue; + dev_info(tas_dev->dev, + "%s: conf %d, block type:%s\t device idx = 0x%02x\n", + __func__, conf_no, + blocktype[cfg_info[conf_no]->blk_data[j] + ->block_type-1], cfg_info[conf_no]->blk_data[j] + ->dev_idx); + + for (k = 0; k < (int)cfg_info[conf_no]->blk_data[j] + ->n_subblks; k++) { + if (cfg_info[conf_no]->blk_data[j]->dev_idx) { + chn = + cfg_info[conf_no]->blk_data[j]->dev_idx + - 1; + chnend = + cfg_info[conf_no]->blk_data[j]->dev_idx; + } else { + chn = 0; + chnend = tas_dev->ndev; + } + for (; chn < chnend; chn++) + tas_dev->tasdevice[chn].is_loading = true; + + rc = tasdevice_process_block(tas_dev, + cfg_info[conf_no]->blk_data[j]->regdata + + length, + cfg_info[conf_no]->blk_data[j]->dev_idx, + cfg_info[conf_no]->blk_data[j]->block_size - + length); + length += rc; + if (cfg_info[conf_no]->blk_data[j]->block_size < + length) { + dev_err(tas_dev->dev, + "%s: ERROR:%u %u out of memory\n", + __func__, length, + cfg_info[conf_no]->blk_data[j] + ->block_size); + break; + } + } + if (length != cfg_info[conf_no]->blk_data[j]->block_size) + dev_err(tas_dev->dev, + "%s: %u %u size is not same\n", + __func__, length, + cfg_info[conf_no]->blk_data[j]->block_size); + + } + +out: + return; +} + +static struct tasdevice_config_info *tasdevice_add_config( + void *pContext, unsigned char *config_data, + unsigned int config_size) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *)pContext; + struct tasdevice_config_info *cfg_info = NULL; + int config_offset = 0, i = 0; + + cfg_info = kzalloc( + sizeof(struct tasdevice_config_info), GFP_KERNEL); + if (!cfg_info) + goto out; + + if (tas_dev->rcabin.fw_hdr.binary_version_num >= 0x105) { + if (config_offset + 64 > (int)config_size) { + dev_err(tas_dev->dev, + "add config: Out of memory\n"); + goto out; + } + config_offset += 64; + } + + if (config_offset + 4 > (int)config_size) { + dev_err(tas_dev->dev, + "add config: Out of memory\n"); + goto out; + } + cfg_info->nblocks = + SMS_HTONL(config_data[config_offset], + config_data[config_offset + 1], + config_data[config_offset + 2], config_data[config_offset + 3]); + config_offset += 4; + + cfg_info->blk_data = kcalloc( + cfg_info->nblocks, sizeof(struct tasdev_blk_data *), + GFP_KERNEL); + if (!cfg_info->blk_data) { + dev_err(tas_dev->dev, + "add config: blk_data alloc failed!\n"); + goto out; + } + cfg_info->real_nblocks = 0; + for (i = 0; i < (int)cfg_info->nblocks; i++) { + if (config_offset + 12 > config_size) { + dev_err(tas_dev->dev, + "add config: Out of memory: i = %d nblocks = %u!\n", + i, cfg_info->nblocks); + break; + } + cfg_info->blk_data[i] = kzalloc( + sizeof(struct tasdev_blk_data), GFP_KERNEL); + if (!cfg_info->blk_data[i]) + break; + + cfg_info->blk_data[i]->dev_idx = config_data[config_offset]; + config_offset++; + + cfg_info->blk_data[i]->block_type = config_data[config_offset]; + config_offset++; + + if (cfg_info->blk_data[i]->block_type == + TASDEVICE_BIN_BLK_PRE_POWER_UP) { + if (cfg_info->blk_data[i]->dev_idx == 0) { + cfg_info->active_dev = 1; + } else { + cfg_info->active_dev = + 1 << + (cfg_info->blk_data[i]->dev_idx - 1); + } + } + cfg_info->blk_data[i]->yram_checksum = + SMS_HTONS(config_data[config_offset], + config_data[config_offset + 1]); + config_offset += 2; + cfg_info->blk_data[i]->block_size = + SMS_HTONL(config_data[config_offset], + config_data[config_offset + 1], + config_data[config_offset + 2], + config_data[config_offset + 3]); + config_offset += 4; + + cfg_info->blk_data[i]->n_subblks = + SMS_HTONL(config_data[config_offset], + config_data[config_offset + 1], + config_data[config_offset + 2], + config_data[config_offset + 3]); + + config_offset += 4; + cfg_info->blk_data[i]->regdata = kzalloc( + cfg_info->blk_data[i]->block_size, GFP_KERNEL); + if (cfg_info->blk_data[i]->regdata == 0) { + dev_err(tas_dev->dev, + "add config: regdata alloc failed!\n"); + goto out; + } + if (config_offset + cfg_info->blk_data[i]->block_size + > config_size) { + dev_err(tas_dev->dev, + "%s: block_size Out of memory: i = %d nblocks = %u!\n", + __func__, i, + cfg_info->nblocks); + break; + } + memcpy(cfg_info->blk_data[i]->regdata, + &config_data[config_offset], + cfg_info->blk_data[i]->block_size); + config_offset += cfg_info->blk_data[i]->block_size; + cfg_info->real_nblocks += 1; + } +out: + return cfg_info; +} +static int tas2781_digital_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int val; + int ret = 0; + + /* Read the primary device as the whole */ + ret = tasdevice_dev_read(tas_dev, 0, mc->reg, &val); + if (ret) { + dev_err(tas_dev->dev, + "%s, get digital vol error\n", + __func__); + goto out; + } + val = (val > mc->max) ? mc->max : val; + val = mc->invert ? mc->max - val : val; + ucontrol->value.integer.value[0] = val; + +out: + return ret; +} + +static int tas2781_digital_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int val; + int i, ret = 0; + + val = ucontrol->value.integer.value[0]; + val = (val > mc->max) ? mc->max : val; + val = mc->invert ? mc->max - val : val; + val = (val < 0) ? 0 : val; + if (tas_dev->set_global_mode != NULL) { + ret = tasdevice_dev_write(tas_dev, tas_dev->ndev, + mc->reg, (unsigned int)val); + if (ret) + dev_err(tas_dev->dev, + "%s, set digital vol error in global mode\n", + __func__); + } else { + for (i = 0; i < tas_dev->ndev; i++) { + ret = tasdevice_dev_write(tas_dev, i, + mc->reg, (unsigned int)val); + if (ret) + dev_err(tas_dev->dev, + "%s, set digital vol error in device %d\n", + __func__, i); + } + } + + return ret; +} + +static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int val; + unsigned char mask = 0; + int ret = 0; + + /* Read the primary device */ + ret = tasdevice_dev_read(tas_dev, 0, mc->reg, &val); + if (ret) { + dev_err(tas_dev->dev, + "%s, get AMP vol error\n", + __func__); + goto out; + } + + mask = (1 << fls(mc->max)) - 1; + mask <<= mc->shift; + val = (val & mask) >> mc->shift; + val = (val > mc->max) ? mc->max : val; + val = mc->invert ? mc->max - val : val; + ucontrol->value.integer.value[0] = (int)val; + +out: + return ret; +} + +static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int val; + int i, ret = 0; + unsigned char mask = 0; + + mask = (1 << fls(mc->max)) - 1; + mask <<= mc->shift; + val = ucontrol->value.integer.value[0]; + val = (val > mc->max) ? mc->max : val; + val = mc->invert ? mc->max - val : val; + val = (val < 0) ? 0 : val; + for (i = 0; i < tas_dev->ndev; i++) { + ret = tasdevice_dev_update_bits(tas_dev, i, + mc->reg, mask, (unsigned int)(val << mc->shift)); + if (ret) + dev_err(tas_dev->dev, + "%s, set AMP vol error in device %d\n", + __func__, i); + } + + return ret; +} + +static const DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); +static const DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); + +static const struct snd_kcontrol_new tas2781_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Amp Gain Volume", TAS2781_AMP_LEVEL, + 1, 0, 20, 0, tas2781_amp_getvol, + tas2781_amp_putvol, amp_vol_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Digital Volume Control", TAS2781_DVC_LVL, + 0, 0, 200, 1, tas2781_digital_getvol, + tas2781_digital_putvol, dvc_tlv), +}; + +static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *p_tasdevice = + snd_soc_component_get_drvdata(codec); + + /* Codec Lock Hold*/ + mutex_lock(&p_tasdevice->codec_lock); + p_tasdevice->rcabin.profile_cfg_id = + ucontrol->value.integer.value[0]; + /* Codec Lock Release*/ + mutex_unlock(&p_tasdevice->codec_lock); + + return 0; +} + +static int tasdevice_info_programs(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *p_tasdevice = + snd_soc_component_get_drvdata(codec); + struct tasdevice_fw *tas_fw = p_tasdevice->fmw; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + /* Codec Lock Hold*/ + mutex_lock(&p_tasdevice->codec_lock); + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = (int)tas_fw->n_programs; + /* Codec Lock Release*/ + mutex_unlock(&p_tasdevice->codec_lock); + return 0; +} + +static int tasdevice_info_configurations( + struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *p_tasdevice = + snd_soc_component_get_drvdata(codec); + struct tasdevice_fw *tas_fw = p_tasdevice->fmw; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + /* Codec Lock Hold*/ + mutex_lock(&p_tasdevice->codec_lock); + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = (int)tas_fw->n_configurations - 1; + /* Codec Lock Release*/ + mutex_unlock(&p_tasdevice->codec_lock); + + return 0; +} + +static int tasdevice_info_profile(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *p_tasdevice = + snd_soc_component_get_drvdata(codec); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + /* Codec Lock Hold*/ + mutex_lock(&p_tasdevice->codec_lock); + uinfo->count = 1; + uinfo->value.integer.min = 1; + uinfo->value.integer.max = max(0, p_tasdevice->rcabin.ncfgs); + /* Codec Lock Release*/ + mutex_unlock(&p_tasdevice->codec_lock); + return 0; +} + +static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *p_tasdevice + = snd_soc_component_get_drvdata(codec); + + /* Codec Lock Hold*/ + mutex_lock(&p_tasdevice->codec_lock); + ucontrol->value.integer.value[0] = + p_tasdevice->rcabin.profile_cfg_id; + /* Codec Lock Release*/ + mutex_unlock(&p_tasdevice->codec_lock); + + return 0; +} + +static int tasdevice_create_controls(struct tasdevice_priv *tas_dev) +{ + int nr_controls = 1, ret = 0, mix_index = 0; + char *name = NULL; + struct snd_kcontrol_new *tasdevice_profile_controls = NULL; + + tasdevice_profile_controls = devm_kzalloc(tas_dev->dev, + nr_controls * sizeof(tasdevice_profile_controls[0]), + GFP_KERNEL); + if (tasdevice_profile_controls == NULL) { + ret = -ENOMEM; + goto out; + } + + /* Create a mixer item for selecting the active profile */ + name = devm_kzalloc(tas_dev->dev, + MAX_CONTROL_NAME, GFP_KERNEL); + if (!name) { + ret = -ENOMEM; + goto out; + } + scnprintf(name, MAX_CONTROL_NAME, "TASDEVICE Profile id"); + tasdevice_profile_controls[mix_index].name = name; + tasdevice_profile_controls[mix_index].iface = + SNDRV_CTL_ELEM_IFACE_MIXER; + tasdevice_profile_controls[mix_index].info = + tasdevice_info_profile; + tasdevice_profile_controls[mix_index].get = + tasdevice_get_profile_id; + tasdevice_profile_controls[mix_index].put = + tasdevice_set_profile_id; + mix_index++; + + ret = snd_soc_add_component_controls(tas_dev->codec, + tasdevice_profile_controls, + nr_controls < mix_index ? nr_controls : mix_index); + + tas_dev->tas_ctrl.nr_controls = + nr_controls < mix_index ? nr_controls : mix_index; +out: + return ret; +} + +static int tasdevice_program_get(struct snd_kcontrol *pKcontrol, + struct snd_ctl_elem_value *pValue) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(pKcontrol); + struct tasdevice_priv *pTAS2781 = snd_soc_component_get_drvdata(codec); + + mutex_lock(&pTAS2781->codec_lock); + pValue->value.integer.value[0] = pTAS2781->cur_prog; + mutex_unlock(&pTAS2781->codec_lock); + return 0; +} + +static int tasdevice_program_put(struct snd_kcontrol *pKcontrol, + struct snd_ctl_elem_value *pValue) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(pKcontrol); + struct tasdevice_priv *pTAS2781 = snd_soc_component_get_drvdata(codec); + unsigned int nProgram = pValue->value.integer.value[0]; + + mutex_lock(&pTAS2781->codec_lock); + pTAS2781->cur_prog = nProgram; + mutex_unlock(&pTAS2781->codec_lock); + return 0; +} + +static int tasdevice_configuration_get( + struct snd_kcontrol *pKcontrol, + struct snd_ctl_elem_value *pValue) +{ + + struct snd_soc_component *codec + = snd_soc_kcontrol_component(pKcontrol); + struct tasdevice_priv *pTAS2781 = snd_soc_component_get_drvdata(codec); + + mutex_lock(&pTAS2781->codec_lock); + pValue->value.integer.value[0] = pTAS2781->cur_conf; + mutex_unlock(&pTAS2781->codec_lock); + return 0; +} + +static int tasdevice_configuration_put( + struct snd_kcontrol *pKcontrol, + struct snd_ctl_elem_value *pValue) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(pKcontrol); + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(codec); + unsigned int n_configuration = pValue->value.integer.value[0]; + + mutex_lock(&tas_dev->codec_lock); + tas_dev->cur_conf = n_configuration; + mutex_unlock(&tas_dev->codec_lock); + return 0; +} + +static int tasdevice_dsp_create_control( + struct tasdevice_priv *tas_dev) +{ + int nr_controls = 2, ret = 0, mix_index = 0; + char *program_name = NULL; + char *configuration_name = NULL; + struct snd_kcontrol_new *tasdevice_dsp_controls = NULL; + + tasdevice_dsp_controls = devm_kzalloc(tas_dev->dev, + nr_controls * sizeof(tasdevice_dsp_controls[0]), + GFP_KERNEL); + if (tasdevice_dsp_controls == NULL) { + ret = -ENOMEM; + goto out; + } + + /* Create a mixer item for selecting the active profile */ + program_name = devm_kzalloc(tas_dev->dev, + MAX_CONTROL_NAME, GFP_KERNEL); + configuration_name = devm_kzalloc(tas_dev->dev, + MAX_CONTROL_NAME, GFP_KERNEL); + if (!program_name || !configuration_name) { + ret = -ENOMEM; + goto out; + } + + scnprintf(program_name, MAX_CONTROL_NAME, "Program"); + tasdevice_dsp_controls[mix_index].name = program_name; + tasdevice_dsp_controls[mix_index].iface = + SNDRV_CTL_ELEM_IFACE_MIXER; + tasdevice_dsp_controls[mix_index].info = + tasdevice_info_programs; + tasdevice_dsp_controls[mix_index].get = + tasdevice_program_get; + tasdevice_dsp_controls[mix_index].put = + tasdevice_program_put; + mix_index++; + + scnprintf(configuration_name, MAX_CONTROL_NAME, "Configuration"); + tasdevice_dsp_controls[mix_index].name = configuration_name; + tasdevice_dsp_controls[mix_index].iface = + SNDRV_CTL_ELEM_IFACE_MIXER; + tasdevice_dsp_controls[mix_index].info = + tasdevice_info_configurations; + tasdevice_dsp_controls[mix_index].get = + tasdevice_configuration_get; + tasdevice_dsp_controls[mix_index].put = + tasdevice_configuration_put; + mix_index++; + + ret = snd_soc_add_component_controls(tas_dev->codec, + tasdevice_dsp_controls, + nr_controls < mix_index ? nr_controls : mix_index); + + tas_dev->tas_ctrl.nr_controls += nr_controls; +out: + return ret; +} + +static void tasdevice_rca_ready(const struct firmware *fmw, + void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + struct tasdevice_rca *rca = NULL; + struct tasdevice_rca_hdr *fw_hdr = NULL; + struct tasdevice_config_info **cfg_info = NULL; + const struct firmware *fw_entry = NULL; + unsigned char *buf = NULL; + int offset = 0, i = 0; + unsigned int total_config_sz = 0; + int ret = 0; + + mutex_lock(&tas_dev->codec_lock); + rca = &(tas_dev->rcabin); + fw_hdr = &(rca->fw_hdr); + if (unlikely(!fmw) || unlikely(!fmw->data)) { + dev_err(tas_dev->dev, + "Failed to read %s, no side - effect on driver running\n", + tas_dev->rca_binaryname); + ret = -1; + goto out; + } + buf = (unsigned char *)fmw->data; + + dev_info(tas_dev->dev, "tasdev: rca_ready start\n"); + fw_hdr->img_sz = SMS_HTONL(buf[offset], buf[offset + 1], + buf[offset + 2], buf[offset + 3]); + offset += 4; + if (fw_hdr->img_sz != fmw->size) { + dev_err(tas_dev->dev, + "File size not match, %d %u", (int)fmw->size, + fw_hdr->img_sz); + ret = -1; + goto out; + } + + fw_hdr->checksum = SMS_HTONL(buf[offset], buf[offset + 1], + buf[offset + 2], buf[offset + 3]); + offset += 4; + fw_hdr->binary_version_num = SMS_HTONL(buf[offset], + buf[offset + 1], buf[offset + 2], buf[offset + 3]); + if (fw_hdr->binary_version_num < 0x103) { + dev_err(tas_dev->dev, + "File version 0x%04x is too low", + fw_hdr->binary_version_num); + ret = -1; + goto out; + } + offset += 4; + fw_hdr->drv_fw_version = SMS_HTONL(buf[offset], buf[offset + 1], + buf[offset + 2], buf[offset + 3]); + offset += 8; + fw_hdr->plat_type = buf[offset]; + offset += 1; + fw_hdr->dev_family = buf[offset]; + offset += 1; + fw_hdr->reserve = buf[offset]; + offset += 1; + fw_hdr->ndev = buf[offset]; + offset += 1; + if (fw_hdr->ndev != tas_dev->ndev) { + dev_err(tas_dev->dev, + "ndev(%u) from rcabin and ndev(%u) from DTS does not match\n", + fw_hdr->ndev, + tas_dev->ndev); + ret = -1; + goto out; + } + if (offset + TASDEVICE_DEVICE_SUM > fw_hdr->img_sz) { + dev_err(tas_dev->dev, + "rca_ready: Out of Memory!\n"); + ret = -1; + goto out; + } + + for (i = 0; i < TASDEVICE_DEVICE_SUM; i++, offset++) + fw_hdr->devs[i] = buf[offset]; + + fw_hdr->nconfig = SMS_HTONL(buf[offset], buf[offset + 1], + buf[offset + 2], buf[offset + 3]); + offset += 4; + dev_info(tas_dev->dev, "nconfig = %u\n", fw_hdr->nconfig); + for (i = 0; i < TASDEVICE_CONFIG_SUM; i++) { + fw_hdr->config_size[i] = SMS_HTONL(buf[offset], + buf[offset + 1], buf[offset + 2], buf[offset + 3]); + offset += 4; + total_config_sz += fw_hdr->config_size[i]; + } + dev_info(tas_dev->dev, + "img_sz = %u total_config_sz = %u offset = %d\n", + fw_hdr->img_sz, total_config_sz, offset); + if (fw_hdr->img_sz - total_config_sz != (unsigned int)offset) { + dev_err(tas_dev->dev, "Bin file error!\n"); + ret = -1; + goto out; + } + cfg_info = kcalloc(fw_hdr->nconfig, + sizeof(struct tasdevice_config_info *), + GFP_KERNEL); + + if (!cfg_info) { + ret = -1; + dev_err(tas_dev->dev, "nconfig Memory alloc failed!\n"); + goto out; + } + rca->cfg_info = cfg_info; + rca->ncfgs = 0; + for (i = 0; i < (int)fw_hdr->nconfig; i++) { + cfg_info[i] = tasdevice_add_config(context, &buf[offset], + fw_hdr->config_size[i]); + if (!cfg_info[i]) { + ret = -1; + dev_err(tas_dev->dev, + "add_config Memory alloc failed!\n"); + break; + } + offset += (int)fw_hdr->config_size[i]; + rca->ncfgs += 1; + } + tasdevice_create_controls(tas_dev); + + tasdevice_dsp_remove(tas_dev); + tasdevice_calbin_remove(tas_dev); + tas_dev->fw_state = TASDEVICE_DSP_FW_PENDING; + scnprintf(tas_dev->coef_binaryname, 64, "%s_coef.bin", + tas_dev->dev_name); + ret = request_firmware(&fw_entry, tas_dev->coef_binaryname, + tas_dev->dev); + if (!ret) { + ret = tasdevice_dspfw_ready(fw_entry, tas_dev); + release_firmware(fw_entry); + fw_entry = NULL; + } else { + tas_dev->fw_state = TASDEVICE_DSP_FW_FAIL; + dev_err(tas_dev->dev, "%s: load %s error\n", __func__, + tas_dev->coef_binaryname); + goto out; + } + tasdevice_dsp_create_control(tas_dev); + + tas_dev->fw_state = TASDEVICE_DSP_FW_ALL_OK; + tas_dev->is_glb_calibrated_data_loaded = true; + for (i = 0; i < tas_dev->ndev; i++) { + scnprintf(tas_dev->cal_binaryname[i], 64, "%s_cal_0x%02x.bin", + tas_dev->dev_name, tas_dev->tasdevice[i].dev_addr); + ret = tas2781_load_calibration(tas_dev, + tas_dev->cal_binaryname[i], i); + if (ret != 0) { + dev_err(tas_dev->dev, + "%s: load %s error, no-side effect\n", + __func__, + tas_dev->cal_binaryname[i]); + ret = 0; + tas_dev->is_glb_calibrated_data_loaded = false; + } + } + +out: + mutex_unlock(&tas_dev->codec_lock); + if (fmw) + release_firmware(fmw); + dev_info(tas_dev->dev, "Firmware init complete\n"); +} + +static void tasdevice_config_info_remove(void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + struct tasdevice_rca *rca = &(tas_dev->rcabin); + struct tasdevice_config_info **cfg_info = rca->cfg_info; + int i = 0, j = 0; + + mutex_lock(&tas_dev->dev_lock); + if (cfg_info) { + for (i = 0; i < rca->ncfgs; i++) { + if (cfg_info[i]) { + for (j = 0; j < (int)cfg_info[i]->real_nblocks; + j++) { + kfree( + cfg_info[i]->blk_data[j]->regdata); + kfree(cfg_info[i]->blk_data[j]); + } + kfree(cfg_info[i]->blk_data); + kfree(cfg_info[i]); + } + } + kfree(cfg_info); + } + mutex_unlock(&tas_dev->dev_lock); +} + +static void tasdevice_enable_irq( + struct tasdevice_priv *tas_dev, bool enable) +{ + if (enable) { + if (tas_dev->irq_info.is_irq_enable) + return; + if (gpio_is_valid(tas_dev->irq_info.irq_gpio)) + enable_irq(tas_dev->irq_info.irq); + tas_dev->irq_info.is_irq_enable = true; + } else { + if (gpio_is_valid(tas_dev->irq_info.irq_gpio)) + disable_irq_nosync(tas_dev->irq_info.irq); + tas_dev->irq_info.is_irq_enable = false; + } +} + +static void tasdevice_set_power_state( + struct tasdevice_priv *tas_dev, int state) +{ + switch (state) { + case 0: + schedule_delayed_work(&tas_dev->tuning_switch_wrk, + msecs_to_jiffies(20)); + break; + default: + if (!(tas_dev->pstream || tas_dev->cstream)) { + if (tas_dev->irq_work_func) + tasdevice_enable_irq(tas_dev, false); + tasdevice_select_cfg_blk(tas_dev, + tas_dev->rcabin.profile_cfg_id, + TASDEVICE_BIN_BLK_PRE_SHUTDOWN); + } + break; + } +} + +static int tasdevice_dapm_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *codec = snd_soc_dapm_to_component(w->dapm); + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(codec); + int stream; + + /* Codec Lock Hold */ + mutex_lock(&tas_dev->codec_lock); + stream = strcmp(w->name, "SPK") ? SNDRV_PCM_STREAM_PLAYBACK : + SNDRV_PCM_STREAM_CAPTURE; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + dev_info(tas_dev->dev, "SND_SOC_DAPM_POST_PMU\n"); + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + tas_dev->pstream = 1; + else + tas_dev->cstream = 1; + tasdevice_set_power_state(tas_dev, 0); + break; + case SND_SOC_DAPM_PRE_PMD: + dev_info(tas_dev->dev, "SND_SOC_DAPM_PRE_PMD\n"); + /* stop DSP only when both playback and capture streams + * are deactivated + */ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + tas_dev->pstream = 0; + else + tas_dev->cstream = 0; + if (tas_dev->pstream != 0 || tas_dev->cstream != 0) + goto out; + tasdevice_set_power_state(tas_dev, 1); + break; + } + /* Codec Lock Release*/ + mutex_unlock(&tas_dev->codec_lock); +out: + return 0; +} + +static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT_E("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, + 0, 0, tasdevice_dapm_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SPK("SPK", tasdevice_dapm_event), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_INPUT("DMIC") +}; + +static const struct snd_soc_dapm_route tasdevice_audio_map[] = { + {"SPK", NULL, "ASI"}, + {"OUT", NULL, "SPK"}, + {"ASI OUT", NULL, "DMIC"} +}; + +static int tasdevice_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *codec = dai->component; + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(codec); + int ret = 0; + + if (tas_dev->fw_state != TASDEVICE_DSP_FW_ALL_OK) { + dev_err(tas_dev->dev, "DSP bin file not loaded\n"); + ret = -EINVAL; + } + return ret; +} + +static int tasdevice_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct tasdevice_priv *tas_dev = snd_soc_dai_get_drvdata(dai); + unsigned int fsrate; + unsigned int slot_width; + int bclk_rate; + int rc = 0; + + dev_info(tas_dev->dev, "%s: %s\n", + __func__, substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + "Playback":"Capture"); + + fsrate = params_rate(params); + switch (fsrate) { + case 48000: + break; + case 44100: + break; + default: + dev_err(tas_dev->dev, + "%s: incorrect sample rate = %u\n", + __func__, fsrate); + rc = -EINVAL; + goto out; + } + + slot_width = params_width(params); + switch (slot_width) { + case 16: + break; + case 20: + break; + case 24: + break; + case 32: + break; + default: + dev_err(tas_dev->dev, + "%s: incorrect slot width = %u\n", + __func__, slot_width); + rc = -EINVAL; + goto out; + } + + bclk_rate = snd_soc_params_to_bclk(params); + if (bclk_rate < 0) { + dev_err(tas_dev->dev, + "%s: incorrect bclk rate = %d\n", + __func__, bclk_rate); + rc = bclk_rate; + goto out; + } + dev_info(tas_dev->dev, + "%s: BCLK rate = %d Channel = %d Sample rate = %u slot width = %u\n", + __func__, bclk_rate, params_channels(params), + fsrate, slot_width); +out: + return rc; +} + +static int tasdevice_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct tasdevice_priv *tas_dev = snd_soc_dai_get_drvdata(codec_dai); + + dev_info(tas_dev->dev, + "%s: clk_id = %d, freq = %u, CLK direction %s\n", + __func__, clk_id, freq, + dir == TASDEVICE_CLK_DIR_OUT ? "OUT":"IN"); + + return 0; +} + +static void tuning_switch_routine(struct work_struct *work) +{ + struct tasdevice_priv *tas_dev = + container_of(work, struct tasdevice_priv, + tuning_switch_wrk.work); + struct tasdevice_fw *tas_fmw = tas_dev->fmw; + int profile_cfg_id = 0; + int is_set_glb_mode = 0; + + mutex_lock(&tas_dev->codec_lock); + + if (tas_fmw) { + if (tas_dev->cur_prog >= tas_fmw->n_programs) + /*bypass all in rca is profile id 0*/ + profile_cfg_id = RCA_CONFIGID_BYPASS_ALL; + else { + /*dsp mode or tuning mode*/ + profile_cfg_id = tas_dev->rcabin.profile_cfg_id; + tas_fmw = tas_dev->fmw; + dev_info(tas_dev->dev, "%s: %s\n", __func__, + tas_fmw->configs[tas_dev->cur_conf].name); + is_set_glb_mode = + tasdevice_select_tuningprm_cfg(tas_dev, + tas_dev->cur_prog, + tas_dev->cur_conf, + profile_cfg_id); + if (tas_dev->set_global_mode != NULL) + tas_dev->set_global_mode(tas_dev); + } + } else + profile_cfg_id = RCA_CONFIGID_BYPASS_ALL; + + tasdevice_select_cfg_blk(tas_dev, profile_cfg_id, + TASDEVICE_BIN_BLK_PRE_POWER_UP); + + if (is_set_glb_mode && tas_dev->set_global_mode) + tas_dev->set_global_mode(tas_dev); + if (tas_dev->irq_work_func) + tasdevice_enable_irq(tas_dev, true); + mutex_unlock(&tas_dev->codec_lock); +} + +static const struct snd_soc_dai_ops tasdevice_dai_ops = { + .startup = tasdevice_startup, + .hw_params = tasdevice_hw_params, + .set_sysclk = tasdevice_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver tasdevice_dai_driver[] = { + { + .name = "tas2781_codec", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TASDEVICE_RATES, + .formats = TASDEVICE_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = TASDEVICE_RATES, + .formats = TASDEVICE_FORMATS, + }, + .ops = &tasdevice_dai_ops, + .symmetric_rate = 1, + }, +}; + +static void tas2781_reset(struct tasdevice_priv *tas_dev) +{ + int ret = 0; + int i = 0; + + for (; i < tas_dev->ndev; i++) { + ret = tasdevice_dev_write(tas_dev, i, + TAS2871_REG_SWRESET, + TAS2871_REG_SWRESET_RESET); + if (ret < 0) { + dev_err(tas_dev->dev, "%s: chn %d reset fail, %d\n", + __func__, i, ret); + continue; + } + } + usleep_range(1000, 1050); +} + +static int tasdevice_codec_probe( + struct snd_soc_component *codec) +{ + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + int ret; + + /* Codec Lock Hold */ + mutex_lock(&tas_dev->codec_lock); + + tas_dev->codec = codec; + scnprintf(tas_dev->rca_binaryname, 64, "%s_rca.bin", + tas_dev->dev_name); + ret = request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, + tas_dev->rca_binaryname, tas_dev->dev, GFP_KERNEL, tas_dev, + tasdevice_rca_ready); + if (ret) + dev_err(tas_dev->dev, + "%s: request_firmware_nowait error:0x%08x\n", + __func__, ret); + + /* Codec Lock Release*/ + mutex_unlock(&tas_dev->codec_lock); + + tas2781_reset(tas_dev); + if (tas_dev->set_global_mode != NULL) + tas_dev->set_global_mode(tas_dev); + + return ret; +} + +static void tasdevice_deinit(void *context) +{ + struct tasdevice_priv *tas_dev = (struct tasdevice_priv *) context; + + tasdevice_config_info_remove(tas_dev); + tasdevice_dsp_remove(tas_dev); + tasdevice_calbin_remove(tas_dev); + tas_dev->fw_state = TASDEVICE_DSP_FW_PENDING; +} + +static void tasdevice_codec_remove( + struct snd_soc_component *codec) +{ + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + /* Codec Lock Hold */ + mutex_lock(&tas_dev->codec_lock); + tasdevice_deinit(tas_dev); + /* Codec Lock Release*/ + mutex_unlock(&tas_dev->codec_lock); +} + +static const struct snd_soc_component_driver + soc_codec_driver_tasdevice = { + .probe = tasdevice_codec_probe, + .remove = tasdevice_codec_remove, + .controls = tas2781_snd_controls, + .num_controls = ARRAY_SIZE(tas2781_snd_controls), + .dapm_widgets = tasdevice_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), + .dapm_routes = tasdevice_audio_map, + .num_dapm_routes = ARRAY_SIZE(tasdevice_audio_map), + .idle_bias_on = 1, + .endianness = 1, +}; + +static void irq_work_routine(struct work_struct *pWork) +{ + struct tasdevice_priv *tas_dev = + container_of(pWork, struct tasdevice_priv, + irq_info.irq_work.work); + + mutex_lock(&tas_dev->codec_lock); + if (tas_dev->runtime_suspend) { + dev_info(tas_dev->dev, "%s, Runtime Suspended\n", __func__); + goto end; + } + /*Logical Layer IRQ function, return is ignored*/ + if (tas_dev->irq_work_func) + tas_dev->irq_work_func(tas_dev); + else + dev_info(tas_dev->dev, + "%s, irq_work_func is NULL\n", __func__); +end: + mutex_unlock(&tas_dev->codec_lock); +} + +static void tas2781_irq_work_func(struct tasdevice_priv *tas_dev) +{ + int rc = 0; + unsigned int reg_val = 0, array_size = 0, i = 0, ndev = 0; + unsigned int int_reg_array[] = { + TAS2781_REG_INT_LTCH0, + TAS2781_REG_INT_LTCH1, + TAS2781_REG_INT_LTCH1_0, + TAS2781_REG_INT_LTCH2, + TAS2781_REG_INT_LTCH3, + TAS2781_REG_INT_LTCH4}; + + tasdevice_enable_irq(tas_dev, false); + + array_size = ARRAY_SIZE(int_reg_array); + + for (ndev = 0; ndev < tas_dev->ndev; ndev++) { + for (i = 0; i < array_size; i++) { + rc = tasdevice_dev_read(tas_dev, + ndev, int_reg_array[i], &reg_val); + if (!rc) + dev_info(tas_dev->dev, + "INT STATUS REG 0x%04x=0x%02x\n", + int_reg_array[i], reg_val); + else + dev_err(tas_dev->dev, + "Read Reg 0x%04x error(rc=%d)\n", + int_reg_array[i], rc); + } + } + +} + +static irqreturn_t tasdevice_irq_handler(int irq, + void *dev_id) +{ + struct tasdevice_priv *tas_dev = (struct tasdevice_priv *)dev_id; + + /* get IRQ status after 100 ms */ + schedule_delayed_work(&tas_dev->irq_info.irq_work, + msecs_to_jiffies(100)); + return IRQ_HANDLED; +} + +static int tasdevice_parse_dt(struct tasdevice_priv *tas_dev) +{ + struct device_node *np = tas_dev->dev->of_node; + int rc = 0, i = 0, ndev = 0; + unsigned int dev_addrs[max_chn]; + struct i2c_client *client = (struct i2c_client *)tas_dev->client; + + ndev = of_property_read_variable_u32_array(np, "ti,audio-slots", + dev_addrs, 0, ARRAY_SIZE(dev_addrs)); + if (ndev <= 0) { + ndev = 1; + dev_addrs[0] = client->addr; + } + for (i = 0; i < ndev; i++) { + tas_dev->tasdevice[i].dev_addr = dev_addrs[i]; + dev_info(tas_dev->dev, "%s=0x%02x", + dts_dev_addr_tag[i], + tas_dev->tasdevice[i].dev_addr); + } + if (ndev > 1) { + rc = of_property_read_u32(np, "ti,broadcast-addr", + &(tas_dev->glb_addr.dev_addr)); + if (rc) { + dev_err(tas_dev->dev, + "Looking up node %s failed %d\n", + np->full_name, rc); + tas_dev->glb_addr.dev_addr = 0; + } + } + + tas_dev->ndev = ndev; + + tas_dev->rst_gpio = of_get_named_gpio(np, "reset-gpios", i); + if (gpio_is_valid(tas_dev->rst_gpio)) { + rc = gpio_request(tas_dev->rst_gpio, "reset"); + if (!rc) { + gpio_direction_output(tas_dev->rst_gpio, 1); + dev_info(tas_dev->dev, "reset-gpio = %d", + tas_dev->rst_gpio); + ndev++; + } else + dev_err(tas_dev->dev, + "%s: Failed to request gpio %d\n", + __func__, tas_dev->rst_gpio); + } else + dev_err(tas_dev->dev, + "Looking up node %s failed %d\n", + np->full_name, + tas_dev->rst_gpio); + + strcpy(tas_dev->dev_name, tasdevice_id[tas_dev->chip_id].name); + + tas_dev->irq_info.irq_gpio = of_get_named_gpio(np, + "interrupts", 0); + if (gpio_is_valid(tas_dev->irq_info.irq_gpio)) { + dev_dbg(tas_dev->dev, "interrupts gpio = %d", + tas_dev->irq_info.irq_gpio); + INIT_DELAYED_WORK(&tas_dev->irq_info.irq_work, + irq_work_routine); + + rc = gpio_request(tas_dev->irq_info.irq_gpio, + "AUDEV-IRQ"); + if (!rc) { + gpio_direction_input( + tas_dev->irq_info.irq_gpio); + + tas_dev->irq_info.irq = + gpio_to_irq( + tas_dev->irq_info.irq_gpio); + dev_info(tas_dev->dev, + "irq = %d\n", + tas_dev->irq_info.irq); + + rc = request_threaded_irq( + tas_dev->irq_info.irq, + tasdevice_irq_handler, + NULL, IRQF_TRIGGER_FALLING| + IRQF_ONESHOT, + SMARTAMP_MODULE_NAME, tas_dev); + if (!rc) + disable_irq_nosync( + tas_dev->irq_info.irq); + else + dev_err(tas_dev->dev, + "request_irq failed, %d\n", + rc); + } else + dev_err(tas_dev->dev, + "%s: GPIO %d request error\n", + __func__, + tas_dev->irq_info.irq_gpio); + } else + dev_err(tas_dev->dev, + "Looking up irq-gpio property in node %s failed %d\n", + np->full_name, + tas_dev->irq_info.irq_gpio); + + if (gpio_is_valid(tas_dev->irq_info.irq_gpio) && rc == 0) { + if (tas_dev->chip_id == TAS2781) + tas_dev->irq_work_func = tas2781_irq_work_func; + else + dev_info(tas_dev->dev, "%s: No match irq_work_func\n", + __func__); + } + + return 0; +} + +static int tasdevice_change_chn_book(struct tasdevice_priv *tas_dev, + enum channel chn, int book) +{ + int n_result = 0; + struct i2c_client *pClient = + (struct i2c_client *)tas_dev->client; + + if (chn < tas_dev->ndev) { + if (tas_dev->glb_addr.ref_cnt != 0) { + tas_dev->glb_addr.ref_cnt = 0; + tas_dev->glb_addr.cur_book = -1; + } + pClient->addr = tas_dev->tasdevice[chn].dev_addr; + if (tas_dev->tasdevice[chn].cur_book != book) { + n_result = regmap_write(tas_dev->regmap, + TASDEVICE_BOOKCTL_PAGE, 0); + if (n_result < 0) { + dev_err(tas_dev->dev, "%s, ERROR, E=%d\n", + __func__, n_result); + goto out; + } + n_result = regmap_write(tas_dev->regmap, + TASDEVICE_BOOKCTL_REG, book); + if (n_result < 0) { + dev_err(tas_dev->dev, "%s, ERROR, E=%d\n", + __func__, n_result); + goto out; + } + tas_dev->tasdevice[chn].cur_book = book; + } + } else if (chn == tas_dev->ndev) { + int i = 0; + + if (tas_dev->glb_addr.ref_cnt == 0) + for (i = 0; i < tas_dev->ndev; i++) + tas_dev->tasdevice[i].cur_book = -1; + pClient->addr = tas_dev->glb_addr.dev_addr; + if (tas_dev->glb_addr.cur_book != book) { + n_result = regmap_write(tas_dev->regmap, + TASDEVICE_BOOKCTL_PAGE, 0); + if (n_result < 0) { + dev_err(tas_dev->dev, + "%s, 0ERROR, E=%d\n", + __func__, n_result); + goto out; + } + n_result = regmap_write(tas_dev->regmap, + TASDEVICE_BOOKCTL_REG, book); + if (n_result < 0) { + dev_err(tas_dev->dev, + "%s, book%xERROR, E=%d\n", + __func__, book, n_result); + goto out; + } + tas_dev->glb_addr.cur_book = book; + } + + tas_dev->glb_addr.ref_cnt++; + } else + dev_err(tas_dev->dev, + "%s, ERROR, no such channel(%d)\n", + __func__, chn); + +out: + return n_result; +} + +int tasdevice_dev_read(struct tasdevice_priv *tas_dev, + enum channel chn, unsigned int reg, unsigned int *pValue) +{ + int n_result = 0; + + mutex_lock(&tas_dev->dev_lock); + if (chn < tas_dev->ndev) { + n_result = tasdevice_change_chn_book(tas_dev, chn, + TASDEVICE_BOOK_ID(reg)); + if (n_result < 0) + goto out; + + n_result = regmap_read(tas_dev->regmap, + TASDEVICE_PGRG(reg), pValue); + if (n_result < 0) + dev_err(tas_dev->dev, "%s, ERROR,E=%d\n", + __func__, n_result); + } else { + + dev_err(tas_dev->dev, "%s, ERROR, no such channel(%d)\n", + __func__, chn); + } + +out: + mutex_unlock(&tas_dev->dev_lock); + return n_result; +} + +int tasdevice_dev_write(struct tasdevice_priv *tas_dev, + enum channel chn, unsigned int reg, unsigned int value) +{ + int n_result = 0; + + mutex_lock(&tas_dev->dev_lock); + if (chn <= tas_dev->ndev) { + n_result = tasdevice_change_chn_book(tas_dev, chn, + TASDEVICE_BOOK_ID(reg)); + if (n_result < 0) + goto out; + + n_result = regmap_write(tas_dev->regmap, + TASDEVICE_PGRG(reg), value); + if (n_result < 0) + dev_err(tas_dev->dev, "%s, ERROR, E=%d\n", + __func__, n_result); + } else + dev_err(tas_dev->dev, "%s, ERROR, no such channel(%d)\n", + __func__, chn); +out: + mutex_unlock(&tas_dev->dev_lock); + return n_result; +} + +int tasdevice_dev_bulk_write( + struct tasdevice_priv *tas_dev, enum channel chn, + unsigned int reg, unsigned char *p_data, + unsigned int n_length) +{ + int n_result = 0; + + mutex_lock(&tas_dev->dev_lock); + if (chn <= tas_dev->ndev) { + n_result = tasdevice_change_chn_book(tas_dev, chn, + TASDEVICE_BOOK_ID(reg)); + if (n_result < 0) + goto out; + + n_result = regmap_bulk_write(tas_dev->regmap, + TASDEVICE_PGRG(reg), p_data, n_length); + if (n_result < 0) + dev_err(tas_dev->dev, "%s, ERROR, E=%d\n", + __func__, n_result); + } else + dev_err(tas_dev->dev, "%s, ERROR, no such channel(%d)\n", + __func__, chn); +out: + mutex_unlock(&tas_dev->dev_lock); + return n_result; +} + +int tasdevice_dev_bulk_read(struct tasdevice_priv *tas_dev, + enum channel chn, unsigned int reg, unsigned char *p_data, + unsigned int n_length) +{ + int n_result = 0; + + mutex_lock(&tas_dev->dev_lock); + if (chn < tas_dev->ndev) { + n_result = tasdevice_change_chn_book(tas_dev, chn, + TASDEVICE_BOOK_ID(reg)); + if (n_result < 0) + goto out; + + n_result = regmap_bulk_read(tas_dev->regmap, + TASDEVICE_PGRG(reg), p_data, n_length); + if (n_result < 0) + dev_err(tas_dev->dev, "%s, ERROR, E=%d\n", + __func__, n_result); + } else + dev_err(tas_dev->dev, "%s, ERROR, no such channel(%d)\n", + __func__, chn); + +out: + mutex_unlock(&tas_dev->dev_lock); + return n_result; +} + +int tasdevice_dev_update_bits( + struct tasdevice_priv *tas_dev, enum channel chn, + unsigned int reg, unsigned int mask, unsigned int value) +{ + int n_result = 0; + struct i2c_client *pClient = + (struct i2c_client *)tas_dev->client; + + mutex_lock(&tas_dev->dev_lock); + if (chn < tas_dev->ndev) { + n_result = tasdevice_change_chn_book(tas_dev, chn, + TASDEVICE_BOOK_ID(reg)); + if (n_result < 0) + goto out; + pClient->addr = tas_dev->tasdevice[chn].dev_addr; + n_result = regmap_update_bits(tas_dev->regmap, + TASDEVICE_PGRG(reg), mask, value); + if (n_result < 0) + dev_err(tas_dev->dev, "%s, ERROR, E=%d\n", + __func__, n_result); + } else { + dev_err(tas_dev->dev, "%s, ERROR, no such channel(%d)\n", + __func__, chn); + n_result = -1; + } + +out: + mutex_unlock(&tas_dev->dev_lock); + return n_result; +} + +static void tas2781_set_global_mode(struct tasdevice_priv *tas_dev) +{ + int i = 0; + int ret = 0; + + for (; i < tas_dev->ndev; i++) { + ret = tasdevice_dev_update_bits(tas_dev, i, + TAS2871_MISC_CFG2, TAS2871_GLOBAL_ADDR_MASK, + TAS2871_GLOBAL_ADDR_ENABLE); + if (ret < 0) { + dev_err(tas_dev->dev, + "%s: chn %d set global fail, %d\n", + __func__, i, ret); + continue; + } + } +} + +static int tasdevice_init(struct tasdevice_priv *tas_dev) +{ + int ret = 0, i = 0; + + tas_dev->cur_prog = -1; + tas_dev->cur_conf = -1; + + for (i = 0; i < tas_dev->ndev; i++) { + tas_dev->tasdevice[i].cur_book = -1; + tas_dev->tasdevice[i].cur_prog = -1; + tas_dev->tasdevice[i].cur_conf = -1; + } + mutex_init(&tas_dev->dev_lock); + if (tas_dev->glb_addr.dev_addr != 0 + && tas_dev->glb_addr.dev_addr < 0x7F) + tas_dev->set_global_mode = tas2781_set_global_mode; + dev_set_drvdata(tas_dev->dev, tas_dev); + + INIT_DELAYED_WORK(&tas_dev->tuning_switch_wrk, + tuning_switch_routine); + + mutex_init(&tas_dev->codec_lock); + ret = devm_snd_soc_register_component(tas_dev->dev, + &soc_codec_driver_tasdevice, + tasdevice_dai_driver, ARRAY_SIZE(tasdevice_dai_driver)); + if (ret) + dev_err(tas_dev->dev, "%s: codec register error:0x%08x\n", + __func__, ret); + + INIT_DELAYED_WORK(&tas_dev->irq_info.irq_work, irq_work_routine); + tas_dev->irq_info.is_irq_enable = false; + + dev_info(tas_dev->dev, "i2c register success\n"); + + return ret; +} + +static void tasdevice_remove(struct tasdevice_priv *tas_dev) +{ + if (gpio_is_valid(tas_dev->rst_gpio)) + gpio_free(tas_dev->rst_gpio); + + if (gpio_is_valid(tas_dev->irq_info.irq_gpio)) { + if (delayed_work_pending(&tas_dev->irq_info.irq_work)) { + dev_info(tas_dev->dev, "cancel IRQ work\n"); + cancel_delayed_work(&tas_dev->irq_info.irq_work); + } + cancel_delayed_work_sync(&tas_dev->irq_info.irq_work); + } + + mutex_destroy(&tas_dev->dev_lock); + mutex_destroy(&tas_dev->codec_lock); + snd_soc_unregister_component(tas_dev->dev); +} + +static int tasdevice_pm_suspend(struct device *dev) +{ + struct tasdevice_priv *tas_dev = dev_get_drvdata(dev); + + mutex_lock(&tas_dev->codec_lock); + tas_dev->runtime_suspend = true; + + if (tas_dev->irq_work_func) { + if (delayed_work_pending(&tas_dev->irq_info.irq_work)) { + dev_dbg(tas_dev->dev, "cancel IRQ work\n"); + cancel_delayed_work_sync(&tas_dev->irq_info.irq_work); + } + } + mutex_unlock(&tas_dev->codec_lock); + return 0; +} + +static int tasdevice_pm_resume(struct device *dev) +{ + struct tasdevice_priv *tas_dev = dev_get_drvdata(dev); + + mutex_lock(&tas_dev->codec_lock); + tas_dev->runtime_suspend = false; + mutex_unlock(&tas_dev->codec_lock); + return 0; +} + +const struct dev_pm_ops tasdevice_pm_ops = { + .suspend = tasdevice_pm_suspend, + .resume = tasdevice_pm_resume +}; + +static int tasdevice_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct tasdevice_priv *tas_dev = NULL; + int ret = 0; + + if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_I2C)) { + dev_err(&i2c->dev, + "%s: I2C check failed\n", __func__); + ret = -ENODEV; + goto out; + } + + tas_dev = devm_kzalloc(&i2c->dev, sizeof(*tas_dev), GFP_KERNEL); + if (!tas_dev) { + ret = -ENOMEM; + goto out; + } + tas_dev->dev = &i2c->dev; + tas_dev->client = (void *)i2c; + tas_dev->chip_id = id->driver_data; + + if (i2c->dev.of_node) + ret = tasdevice_parse_dt(tas_dev); + else { + dev_err(tas_dev->dev, "No DTS info\n"); + goto out; + } + + tas_dev->regmap = devm_regmap_init_i2c(i2c, + &tasdevice_regmap); + if (IS_ERR(tas_dev->regmap)) { + ret = PTR_ERR(tas_dev->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + ret = tasdevice_init(tas_dev); + +out: + if (ret < 0 && tas_dev != NULL) + tasdevice_remove(tas_dev); + return ret; + +} + +static void tasdevice_i2c_remove(struct i2c_client *pClient) +{ + struct tasdevice_priv *tas_dev = i2c_get_clientdata(pClient); + + if (tas_dev) + tasdevice_remove(tas_dev); + +} + +static struct i2c_driver tasdevice_i2c_driver = { + .driver = { + .name = "tas2781-codec", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tasdevice_of_match), + .pm = &tasdevice_pm_ops, + }, + .probe = tasdevice_i2c_probe, + .remove = tasdevice_i2c_remove, + .id_table = tasdevice_id, +}; + +module_i2c_driver(tasdevice_i2c_driver); + +MODULE_AUTHOR("Shenghao Ding <shenghao-ding(a)ti.com>"); +MODULE_AUTHOR("Kevin Lu <kevin-lu(a)ti.com>"); +MODULE_DESCRIPTION("ASoC TAS2781 Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2781.h b/sound/soc/codecs/tas2781.h new file mode 100644 index 000000000000..ce9d3d162aa0 --- /dev/null +++ b/sound/soc/codecs/tas2781.h @@ -0,0 +1,176 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier + * + * Copyright (C) 2022 - 2023 Texas Instruments Incorporated + * https://www.ti.com + * + * The TAS2781 driver implements a flexible and configurable algo coff setting + * for one, two, even multiple TAS2781 chips. + * + * Author: Shenghao Ding <shenghao-ding(a)ti.com> + * Kevin Lu <kevin-lu(a)ti.com> + */ + +#ifndef __TAS2781_H__ +#define __TAS2781_H__ + +#include "tas2781-dsp.h" + +#define SMARTAMP_MODULE_NAME "tas2781" +#define TASDEVICE_RETRY_COUNT 3 +#define TASDEVICE_ERROR_FAILED -2 + +#define TASDEVICE_RATES (SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_88200) +#define TASDEVICE_MAX_CHANNELS 8 + +#define TASDEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/*PAGE Control Register (available in page0 of each book) */ +#define TASDEVICE_PAGE_SELECT 0x00 +#define TASDEVICE_BOOKCTL_PAGE 0x00 +#define TASDEVICE_BOOKCTL_REG 127 +#define TASDEVICE_BOOK_ID(reg) (reg / (256 * 128)) +#define TASDEVICE_PAGE_ID(reg) ((reg % (256 * 128)) / 128) +#define TASDEVICE_PAGE_REG(reg) ((reg % (256 * 128)) % 128) +#define TASDEVICE_PGRG(reg) ((reg % (256 * 128))) +#define TASDEVICE_REG(book, page, reg) (((book * 256 * 128) + \ + (page * 128)) + reg) + + /*Software Reset */ +#define TAS2871_REG_SWRESET TASDEVICE_REG(0x0, 0X0, 0x02) +#define TAS2871_REG_SWRESET_RESET BIT(0) + + /* Enable Global addresses */ +#define TAS2871_MISC_CFG2 TASDEVICE_REG(0x0, 0X0, 0x07) +#define TAS2871_GLOBAL_ADDR_MASK BIT(1) +#define TAS2871_GLOBAL_ADDR_ENABLE BIT(1) + + /*I2C Checksum */ +#define TASDEVICE_I2CChecksum TASDEVICE_REG(0x0, 0x0, 0x7E) + + /* Volume control */ +#define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1A) +#define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) +#define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) + +#define TASDEVICE_CMD_SING_W 0x1 +#define TASDEVICE_CMD_BURST 0x2 +#define TASDEVICE_CMD_DELAY 0x3 +#define TASDEVICE_CMD_FIELD_W 0x4 + +enum audio_device { + TAS2781 = 0, +}; + +#define SMS_HTONS(a, b) ((((a)&0x00FF)<<8) | ((b)&0x00FF)) +#define SMS_HTONL(a, b, c, d) ((((a)&0x000000FF)<<24) | \ + (((b)&0x000000FF)<<16) | (((c)&0x000000FF)<<8) | \ + ((d)&0x000000FF)) + +struct tasdevice { + unsigned int dev_addr; + unsigned int err_code; + short cur_prog; + short cur_conf; + bool is_loading; + bool is_loaderr; + bool is_calibrated_data_loaded; + unsigned char cur_book; + struct tasdevice_fw *cali_data_fmw; +}; + +/* + * This item is used to store the generic i2c address of + * all the tas2781 devices for I2C broadcast during the multi-device + * writes, useless in mono case. + */ +struct global_addr { + unsigned char cur_book; + unsigned int dev_addr; + int ref_cnt; +}; + +struct tas_control { + struct snd_kcontrol_new *tasdevice_profile_controls; + int nr_controls; +}; + +struct tasdevice_irqinfo { + int irq_gpio; + int irq; + struct delayed_work irq_work; + bool is_irq_enable; +}; + +struct tasdevice_priv { + struct device *dev; + void *client; + struct regmap *regmap; + struct mutex codec_lock; + struct mutex dev_lock; + int rst_gpio; + struct tasdevice tasdevice[max_chn]; + struct tasdevice_fw *fmw; + struct tasdevice_rca rcabin; + struct tasdevice_irqinfo irq_info; + struct tas_control tas_ctrl; + struct global_addr glb_addr; + int cur_prog; + int cur_conf; + unsigned int chip_id; + void (*set_global_mode)(struct tasdevice_priv *tas_dev); + int (*fw_parse_variable_header)(struct tasdevice_priv *tas_dev, + const struct firmware *fmw, int offset); + int (*fw_parse_program_data)(struct tasdevice_priv *tas_dev, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); + int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_dev, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); + int (*tasdevice_load_block)(struct tasdevice_priv *tas_dev, + struct tasdev_blk *pBlock); + int (*fw_parse_calibration_data)(struct tasdevice_priv *tas_dev, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); + void (*irq_work_func)(struct tasdevice_priv *tas_dev); + int fw_state; + unsigned int magic_num; + unsigned char ndev; + unsigned char dev_name[32]; + unsigned char rca_binaryname[64]; + unsigned char coef_binaryname[64]; + unsigned char cal_binaryname[max_chn][64]; + bool runtime_suspend; + struct delayed_work tuning_switch_wrk; + void *codec; + int sysclk; + int pstream; + int cstream; + bool is_glb_calibrated_data_loaded; +}; + +int tasdevice_dev_read(struct tasdevice_priv *tasdevice, + enum channel chn, unsigned int reg, unsigned int *value); +int tasdevice_process_block(void *context, + unsigned char *data, unsigned char dev_idx, int sublocksize); +int tasdevice_dev_write(struct tasdevice_priv *tasdevice, + enum channel chn, unsigned int reg, unsigned int value); + +int tasdevice_dev_bulk_write( + struct tasdevice_priv *tasdevice, enum channel chn, + unsigned int reg, unsigned char *p_data, unsigned int n_length); + +int tasdevice_dev_bulk_read(struct tasdevice_priv *tasdevice, + enum channel chn, unsigned int reg, unsigned char *p_data, + unsigned int n_length); + +int tasdevice_dev_update_bits( + struct tasdevice_priv *tasdevice, enum channel chn, + unsigned int reg, unsigned int mask, unsigned int value); + +#endif /*__TAS2781_H__ */ -- 2.34.1

1 0

[PATCH] ASoC: Intel: bytcr_rt5640: Add quirk for the Acer Iconia One 7 B1-750
by Hans de Goede 17 Feb '23

17 Feb '23

The Acer Iconia One 7 B1-750 tablet mostly works fine with the defaults for an Bay Trail CR tablet. Except for the internal mic, instead of an analog mic on IN3 a digital mic on DMIC1 is uses. Add a quirk with these settings for this tablet. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 79e0039c79a3..5a12940ef907 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -533,6 +533,18 @@ static int byt_rt5640_aif1_hw_params(struct snd_pcm_substream *substream, /* Please keep this list alphabetically sorted */ static const struct dmi_system_id byt_rt5640_quirk_table[] = { + { /* Acer Iconia One 7 B1-750 */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Insyde"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "VESPA2"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD1_IN4P | + BYT_RT5640_OVCD_TH_1500UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* Acer Iconia Tab 8 W1-810 */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Acer"), -- 2.39.1

2 1

[PATCH] skl-topology_3627: using sprintf changed to snprintf
by Natalia Petrova 17 Feb '23

17 Feb '23

Use snprintf instead of sprintf to avoid buffer overflow. Found by Linux Verification Center (linuxtesting.org) with SVACE. Fixes: 2d744ecf2b98 ("ASoC: Intel: Skylake: Automatic DMIC format configuration according to information from NHLT") Signed-off-by: Natalia Petrova <n.petrova(a)fintech.ru> --- sound/soc/intel/skylake/skl-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index e06eac592da1..ac06926e651d 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3660,7 +3660,7 @@ static int skl_tplg_complete(struct snd_soc_component *component) se = (struct soc_enum *)kcontrol->private_value; texts = dobj->control.dtexts; - sprintf(chan_text, "c%d", mach->mach_params.dmic_num); + snprintf(chan_text, sizeof(chan_text), "c%d", mach->mach_params.dmic_num); for (i = 0; i < se->items; i++) { if (strstr(texts[i], chan_text)) { -- 2.34.1

1 0

[PATCH v2 0/9] Refactor Vangogh acp5x machine driver
by Lucas Tanure 17 Feb '23

17 Feb '23

Provide small fixes and refactor the code for easier insertion of a new platform using the same acp5x machine driver. Changes since V1: - Don't use smaller variable names - Don't use 100 limit for all lines - Use component prefixes to avoid collisions - Improved commit messages Lucas Tanure (9): ASoC: amd: vangogh: Remove unnecessary init function ASoC: amd: vangogh: Small code refactor ASoC: amd: vangogh: use sizeof of variable instead of struct type ASoC: amd: vangogh: remove unnecessarily included headers ASoC: amd: vangogh: use for_each_rtd_components instead of for ASoC: amd: vangogh: Check Bit Clock rate before snd_soc_dai_set_pll ASoC: amd: vangogh: Move nau8821 and CPU side code up for future platform ASoC: amd: vangogh: Centralize strings definition ASoC: amd: vangogh: Add components prefix in structs and function names sound/soc/amd/vangogh/acp5x-mach.c | 297 ++++++++++++++--------------- 1 file changed, 144 insertions(+), 153 deletions(-) -- 2.39.2

1 9

[PATCH 0/3] ASoC: mchp-pdmc: fix poc noises when starting capture
by Claudiu Beznea 17 Feb '23

17 Feb '23

To start capture on Microchip PDMC the enable bits for each supported microphone need to be set. After this bit is set the PDMC starts to receive data from microphones and it considers this data as valid data. Thus if microphones are not ready the PDMC captures anyway data from its lines. This data is interpreted by the human ear as poc noises. To avoid this the following software workaround need to be applied when starting capture: 1/ enable PDMC channel 2/ wait 150ms 3/ execute 16 dummy reads from RHR 4/ clear interrupts 5/ enable interrupts 6/ enable DMA channel For this workaround to work step 6 need to be executed at the end. For step 6 was added patch 1/3 from this series. With this, driver sets its struct snd_dmaengine_pcm_config::start_dma_last = 1 and proper action is taken based on this flag when starting DAI trigger vs DMA. The other solution that was identified for this was to extend the already existing mechanism around struct snd_soc_dai_link::stop_dma_first. The downside of this was that a potential struct snd_soc_dai_link::start_dma_last would have to be populated on sound card driver thus, had to be taken into account in all sound card drivers. At the moment, the mchp-pdmc is used only with simple-audio-card. In case of simple-audio-card a new DT binding would had to be introduced to specify this action on dai-link descriptions (as of my investigation). Please advice what might be the best approach. Thank you, Claudiu Beznea Claudiu Beznea (3): ASoC: soc-generic-dmaengine-pcm: add option to start DMA after DAI ASoC: dt-bindings: sama7g5-pdmc: add microchip,startup-delay-us binding ASoC: mchp-pdmc: fix poc noise at capture startup .../sound/microchip,sama7g5-pdmc.yaml | 6 ++ include/sound/dmaengine_pcm.h | 1 + include/sound/soc-component.h | 2 + sound/soc/atmel/mchp-pdmc.c | 55 +++++++++++++++++-- sound/soc/soc-generic-dmaengine-pcm.c | 8 ++- sound/soc/soc-pcm.c | 27 +++++++-- 6 files changed, 86 insertions(+), 13 deletions(-) -- 2.34.1

5 14

[PATCH] ASoC: nau8821: Implement DRC controls
by Seven Lee 17 Feb '23

17 Feb '23

This patch is support dynamic range compression controls. Signed-off-by: Seven Lee <wtli(a)nuvoton.com> Signed-off-by: Seven Lee <scott6986(a)gmail.com> --- sound/soc/codecs/nau8821.c | 96 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/nau8821.h | 23 +++++++++ 2 files changed, 119 insertions(+) diff --git a/sound/soc/codecs/nau8821.c b/sound/soc/codecs/nau8821.c index 4a72b94e8410..fee970427a24 100644 --- a/sound/soc/codecs/nau8821.c +++ b/sound/soc/codecs/nau8821.c @@ -322,12 +322,92 @@ static const struct soc_enum nau8821_dac_oversampl_enum = SOC_ENUM_SINGLE(NAU8821_R2C_DAC_CTRL1, NAU8821_DAC_OVERSAMPLE_SFT, ARRAY_SIZE(nau8821_dac_oversampl), nau8821_dac_oversampl); +static const char * const nau8821_adc_drc_noise_gate[] = { + "1:1", "2:1", "4:1", "8:1" }; + +static const struct soc_enum nau8821_adc_drc_noise_gate_enum = + SOC_ENUM_SINGLE(NAU8821_R38_ADC_DRC_SLOPES, NAU8821_DRC_NG_SLP_ADC_SFT, + ARRAY_SIZE(nau8821_adc_drc_noise_gate), + nau8821_adc_drc_noise_gate); + +static const char * const nau8821_adc_drc_expansion_slope[] = { + "1:1", "2:1", "4:1" }; + +static const struct soc_enum nau8821_adc_drc_expansion_slope_enum = + SOC_ENUM_SINGLE(NAU8821_R38_ADC_DRC_SLOPES, NAU8821_DRC_EXP_SLP_ADC_SFT, + ARRAY_SIZE(nau8821_adc_drc_expansion_slope), + nau8821_adc_drc_expansion_slope); + +static const char * const nau8821_adc_drc_lower_region[] = { + "0", "1:2", "1:4", "1:8", "1:16", "", "", "1:1" }; + +static const struct soc_enum nau8821_adc_drc_lower_region_enum = + SOC_ENUM_SINGLE(NAU8821_R38_ADC_DRC_SLOPES, + NAU8821_DRC_CMP2_SLP_ADC_SFT, + ARRAY_SIZE(nau8821_adc_drc_lower_region), + nau8821_adc_drc_lower_region); + +static const char * const nau8821_higher_region[] = { + "0", "1:2", "1:4", "1:8", "1:16", "", "", "1:1" }; + +static const struct soc_enum nau8821_higher_region_enum = + SOC_ENUM_SINGLE(NAU8821_R38_ADC_DRC_SLOPES, + NAU8821_DRC_CMP1_SLP_ADC_SFT, + ARRAY_SIZE(nau8821_higher_region), + nau8821_higher_region); + +static const char * const nau8821_limiter_slope[] = { + "0", "1:2", "1:4", "1:8", "1:16", "1:32", "1:64", "1:1" }; + +static const struct soc_enum nau8821_limiter_slope_enum = + SOC_ENUM_SINGLE(NAU8821_R38_ADC_DRC_SLOPES, + NAU8821_DRC_LMT_SLP_ADC_SFT, ARRAY_SIZE(nau8821_limiter_slope), + nau8821_limiter_slope); + +static const char * const nau8821_detection_attack_time[] = { + "Ts", "3Ts", "7Ts", "15Ts", "31Ts", "63Ts", "127Ts", "255Ts", + "", "511Ts" }; + +static const struct soc_enum nau8821_detection_attack_time_enum = + SOC_ENUM_SINGLE(NAU8821_R39_ADC_DRC_ATKDCY, + NAU8821_DRC_PK_COEF1_ADC_SFT, + ARRAY_SIZE(nau8821_detection_attack_time), + nau8821_detection_attack_time); + +static const char * const nau8821_detection_release_time[] = { + "63Ts", "127Ts", "255Ts", "511Ts", "1023Ts", "2047Ts", "4095Ts", + "8191Ts", "", "16383Ts" }; + +static const struct soc_enum nau8821_detection_release_time_enum = + SOC_ENUM_SINGLE(NAU8821_R39_ADC_DRC_ATKDCY, + NAU8821_DRC_PK_COEF2_ADC_SFT, + ARRAY_SIZE(nau8821_detection_release_time), + nau8821_detection_release_time); + +static const char * const nau8821_attack_time[] = { + "Ts", "3Ts", "7Ts", "15Ts", "31Ts", "63Ts", "127Ts", "255Ts", + "511Ts", "1023Ts", "2047Ts", "4095Ts", "8191Ts" }; + +static const struct soc_enum nau8821_attack_time_enum = + SOC_ENUM_SINGLE(NAU8821_R39_ADC_DRC_ATKDCY, NAU8821_DRC_ATK_ADC_SFT, + ARRAY_SIZE(nau8821_attack_time), nau8821_attack_time); + +static const char * const nau8821_decay_time[] = { + "63Ts", "127Ts", "255Ts", "511Ts", "1023Ts", "2047Ts", "4095Ts", + "8191Ts", "16383Ts", "32757Ts", "65535Ts" }; + +static const struct soc_enum nau8821_decay_time_enum = + SOC_ENUM_SINGLE(NAU8821_R39_ADC_DRC_ATKDCY, NAU8821_DRC_DCY_ADC_SFT, + ARRAY_SIZE(nau8821_decay_time), nau8821_decay_time); + static const DECLARE_TLV_DB_MINMAX_MUTE(adc_vol_tlv, -6600, 2400); static const DECLARE_TLV_DB_MINMAX_MUTE(sidetone_vol_tlv, -4200, 0); static const DECLARE_TLV_DB_MINMAX(hp_vol_tlv, -900, 0); static const DECLARE_TLV_DB_SCALE(playback_vol_tlv, -6600, 50, 1); static const DECLARE_TLV_DB_MINMAX(fepga_gain_tlv, -100, 3600); static const DECLARE_TLV_DB_MINMAX_MUTE(crosstalk_vol_tlv, -7000, 2400); +static const DECLARE_TLV_DB_MINMAX(drc_knee4_tlv, -9800, -3500); +static const DECLARE_TLV_DB_MINMAX(drc_knee3_tlv, -8100, -1800); static const struct snd_kcontrol_new nau8821_controls[] = { SOC_DOUBLE_TLV("Mic Volume", NAU8821_R35_ADC_DGAIN_CTRL1, @@ -346,6 +426,22 @@ static const struct snd_kcontrol_new nau8821_controls[] = { SOC_DOUBLE_TLV("Headphone Crosstalk Volume", NAU8821_R2F_DAC_DGAIN_CTRL, 0, 8, 0xff, 0, crosstalk_vol_tlv), + SOC_SINGLE_TLV("ADC DRC KNEE4", NAU8821_R37_ADC_DRC_KNEE_IP34, + NAU8821_DRC_KNEE4_IP_ADC_SFT, 0x3f, 1, drc_knee4_tlv), + SOC_SINGLE_TLV("ADC DRC KNEE3", NAU8821_R37_ADC_DRC_KNEE_IP34, + NAU8821_DRC_KNEE3_IP_ADC_SFT, 0x3f, 1, drc_knee3_tlv), + + SOC_ENUM("ADC DRC Noise Gate", nau8821_adc_drc_noise_gate_enum), + SOC_ENUM("ADC DRC Expansion Slope", nau8821_adc_drc_expansion_slope_enum), + SOC_ENUM("ADC DRC Lower Region", nau8821_adc_drc_lower_region_enum), + SOC_ENUM("ADC DRC Higher Region", nau8821_higher_region_enum), + SOC_ENUM("ADC DRC Limiter Slope", nau8821_limiter_slope_enum), + SOC_ENUM("ADC DRC Peak Detection Attack Time", nau8821_detection_attack_time_enum), + SOC_ENUM("ADC DRC Peak Detection Release Time", nau8821_detection_release_time_enum), + SOC_ENUM("ADC DRC Attack Time", nau8821_attack_time_enum), + SOC_ENUM("ADC DRC Decay Time", nau8821_decay_time_enum), + SOC_SINGLE("DRC Enable Switch", NAU8821_R36_ADC_DRC_KNEE_IP12, + NAU8821_DRC_ENA_ADC_SFT, 1, 0), SOC_ENUM("ADC Decimation Rate", nau8821_adc_decimation_enum), SOC_ENUM("DAC Oversampling Rate", nau8821_dac_oversampl_enum), diff --git a/sound/soc/codecs/nau8821.h b/sound/soc/codecs/nau8821.h index c44251f54d48..d962293c218e 100644 --- a/sound/soc/codecs/nau8821.h +++ b/sound/soc/codecs/nau8821.h @@ -350,6 +350,29 @@ #define NAU8821_ADCL_CH_VOL_SFT 0 #define NAU8821_ADCL_CH_VOL_MASK 0xff +/* ADC_DRC_KNEE_IP12 (0x36) */ +#define NAU8821_DRC_ENA_ADC_SFT 15 +#define NAU8821_DRC_ENA_ADC_EN (0x1 << NAU8821_DRC_ENA_ADC_SFT) + +/* ADC_DRC_KNEE_IP34 (0x37) */ +#define NAU8821_DRC_KNEE4_IP_ADC_SFT 8 +#define NAU8821_DRC_KNEE4_IP_ADC_MASK (0xff << NAU8821_DRC_KNEE4_IP_ADC_SFT) +#define NAU8821_DRC_KNEE3_IP_ADC_SFT 0 +#define NAU8821_DRC_KNEE3_IP_ADC_MASK 0xff + +/* ADC_DRC_SLOPES (0x38) */ +#define NAU8821_DRC_NG_SLP_ADC_SFT 12 +#define NAU8821_DRC_EXP_SLP_ADC_SFT 9 +#define NAU8821_DRC_CMP2_SLP_ADC_SFT 6 +#define NAU8821_DRC_CMP1_SLP_ADC_SFT 3 +#define NAU8821_DRC_LMT_SLP_ADC_SFT 0 + +/* ADC_DRC_ATKDCY (0x39) */ +#define NAU8821_DRC_PK_COEF1_ADC_SFT 12 +#define NAU8821_DRC_PK_COEF2_ADC_SFT 8 +#define NAU8821_DRC_ATK_ADC_SFT 4 +#define NAU8821_DRC_DCY_ADC_SFT 0 + /* BIQ1_COF10 (0x4a) */ #define NAU8821_BIQ1_DAC_EN_SFT 3 #define NAU8821_BIQ1_DAC_EN_EN (0x1 << NAU8821_BIQ1_DAC_EN_SFT) -- 2.25.1

1 0

[GIT PULL] ASoC fixes for v6.2-rc8-2
by Mark Brown 17 Feb '23

17 Feb '23

The following changes since commit c173ee5b2fa6195066674d66d1d7e191010fb1ff: ASoC: topology: Return -ENOMEM on memory allocation failure (2023-02-07 14:06:26 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.2-rc8-2 for you to fetch changes up to 1f810d2b6b2fbdc5279644d8b2c140b1f7c9d43d: ASoC: SOF: Intel: hda-dai: fix possible stream_tag leak (2023-02-16 16:31:15 +0000) ---------------------------------------------------------------- ASoC: One more fix for v6.2 One more fix from Peter which he'd very much like to get into v6.2. ---------------------------------------------------------------- Pierre-Louis Bossart (1): ASoC: SOF: Intel: hda-dai: fix possible stream_tag leak Rander Wang (1): ASoC: SOF: ops: refine parameters order in function snd_sof_dsp_update8 sound/soc/sof/intel/hda-dai.c | 8 ++++---- sound/soc/sof/ops.h | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-)

2 1

[PATCH] ASoC: mediatek: mt8186: Use devm_platform_ioremap_resource()
by ye.xingchen＠zte.com.cn 17 Feb '23

17 Feb '23

From: Ye Xingchen <ye.xingchen(a)zte.com.cn> Convert platform_get_resource(), devm_ioremap_resource() to a single call to Use devm_platform_ioremap_resource(), as this is exactly what this function does. Signed-off-by: Ye Xingchen <ye.xingchen(a)zte.com.cn> --- sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c b/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c index 41172a82103e..96be444d1000 100644 --- a/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c +++ b/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c @@ -2815,7 +2815,6 @@ static int mt8186_afe_pcm_dev_probe(struct platform_device *pdev) { struct mtk_base_afe *afe; struct mt8186_afe_private *afe_priv; - struct resource *res; struct reset_control *rstc; struct device *dev = &pdev->dev; int i, ret, irq_id; @@ -2836,8 +2835,7 @@ static int mt8186_afe_pcm_dev_probe(struct platform_device *pdev) afe_priv = afe->platform_priv; afe->dev = &pdev->dev; - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - afe->base_addr = devm_ioremap_resource(dev, res); + afe->base_addr = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(afe->base_addr)) return PTR_ERR(afe->base_addr); -- 2.25.1

1 0

[PATCH v2] ASoC: SOF: Intel: hda-dai: fix possible stream_tag leak
by Peter Ujfalusi 17 Feb '23

17 Feb '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The HDaudio stream allocation is done first, and in a second step the LOSIDV parameter is programmed for the multi-link used by a codec. This leads to a possible stream_tag leak, e.g. if a DisplayAudio link is not used. This would happen when a non-Intel graphics card is used and userspace unconditionally uses the Intel Display Audio PCMs without checking if they are connected to a receiver with jack controls. We should first check that there is a valid multi-link entry to configure before allocating a stream_tag. This change aligns the dma_assign and dma_cleanup phases. Complements: b0cd60f3e9f5 ("ALSA/ASoC: hda: clarify bus_get_link() and bus_link_get() helpers") Link: https://github.com/thesofproject/linux/issues/4151 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- Hi Mark, I have added a Complements tag for now to force it follow the rename patch. I need to check how far this needs to go back and I will send a backported version later on. b0cd60f3e9f5 is in 6.2-rc Thanks, Peter sound/soc/sof/intel/hda-dai.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 193b3e74820a..8d9c38d562d3 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -216,6 +216,10 @@ static int hda_link_dma_hw_params(struct snd_pcm_substream *substream, sdev = snd_soc_component_get_drvdata(cpu_dai->component); bus = sof_to_bus(sdev); + hlink = snd_hdac_ext_bus_get_hlink_by_name(bus, codec_dai->component->name); + if (!hlink) + return -EINVAL; + hext_stream = snd_soc_dai_get_dma_data(cpu_dai, substream); if (!hext_stream) { hext_stream = hda_link_stream_assign(bus, substream); @@ -225,10 +229,6 @@ static int hda_link_dma_hw_params(struct snd_pcm_substream *substream, snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)hext_stream); } - hlink = snd_hdac_ext_bus_get_hlink_by_name(bus, codec_dai->component->name); - if (!hlink) - return -EINVAL; - /* set the hdac_stream in the codec dai */ snd_soc_dai_set_stream(codec_dai, hdac_stream(hext_stream), substream->stream); -- 2.39.2

2 1

[PATCH] soc:amd:yp:Add OMEN by HP Gaming Laptop 16z-n000 to quirks
by Joseph Hunkeler 17 Feb '23

17 Feb '23

Enables display microphone on the HP OMEN 16z-n000 (8A42) laptop Signed-off-by: Joseph Hunkeler <jhunkeler(a)gmail.com> --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 363147539..4e681e9c0 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -255,6 +255,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "15NBC1011"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "HP"), + DMI_MATCH(DMI_PRODUCT_NAME, "OMEN by HP Gaming Laptop 16z-n000"), + } + }, {} }; base-commit: e1c04510f521e853019afeca2a5991a5ef8d6a5b -- 2.39.1

2 5

[PATCH] ASoC: SOF: Intel: hda-dai: fix possible stream_tag leak
by Peter Ujfalusi 17 Feb '23

17 Feb '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The HDaudio stream allocation is done first, and in a second step the LOSIDV parameter is programmed for the multi-link used by a codec. This leads to a possible stream_tag leak, e.g. if a DisplayAudio link is not used. This would happen when a non-Intel graphics card is used and userspace unconditionally uses the Intel Display Audio PCMs without checking if they are connected to a receiver with jack controls. We should first check that there is a valid multi-link entry to configure before allocating a stream_tag. This change aligns the dma_assign and dma_cleanup phases. Link: https://github.com/thesofproject/linux/issues/4151 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- Hi Mark, is there still time to send this for v6.2? If not, then I will re-send it with stable 6.2 tag. In fact similar change should be sent to stable kernels but due to a function name change in 6.2 [1], this patch is not going to apply anything older.. [1] b0cd60f3e9f5 ("ALSA/ASoC: hda: clarify bus_get_link() and bus_link_get() helpers") Did a snd_hdac_ext_bus_get_link() -> snd_hdac_ext_bus_get_hlink_by_name() rename I'm not sure if I should add the stable tag and let the stable guys figure it out, but I feel guilty to do so... Thanks, Peter sound/soc/sof/intel/hda-dai.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 193b3e74820a..8d9c38d562d3 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -216,6 +216,10 @@ static int hda_link_dma_hw_params(struct snd_pcm_substream *substream, sdev = snd_soc_component_get_drvdata(cpu_dai->component); bus = sof_to_bus(sdev); + hlink = snd_hdac_ext_bus_get_hlink_by_name(bus, codec_dai->component->name); + if (!hlink) + return -EINVAL; + hext_stream = snd_soc_dai_get_dma_data(cpu_dai, substream); if (!hext_stream) { hext_stream = hda_link_stream_assign(bus, substream); @@ -225,10 +229,6 @@ static int hda_link_dma_hw_params(struct snd_pcm_substream *substream, snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)hext_stream); } - hlink = snd_hdac_ext_bus_get_hlink_by_name(bus, codec_dai->component->name); - if (!hlink) - return -EINVAL; - /* set the hdac_stream in the codec dai */ snd_soc_dai_set_stream(codec_dai, hdac_stream(hext_stream), substream->stream); -- 2.39.2

2 3

[PATCH -next] ASoC: soc-ac97: fix wrong pointer pass to PTR_ERR()
by Yang Yingliang 17 Feb '23

17 Feb '23

Fix wrong pointer pass to PTR_ERR() if devm_gpiod_get_index() fails. Fixes: 3ee0d39c50dc ("ASoC: soc-ac97: Convert to agnostic GPIO API") Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/soc-ac97.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index 40051f513c6a..4e4fe29ade50 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -336,12 +336,12 @@ static int snd_soc_ac97_parse_pinctl(struct device *dev, cfg->sdata_gpio = devm_gpiod_get_index(dev, "ac97", 1, GPIOD_ASIS); if (IS_ERR(cfg->sdata_gpio)) - return dev_err_probe(dev, PTR_ERR(cfg->sync_gpio), "Can't find ac97-sdata gpio\n"); + return dev_err_probe(dev, PTR_ERR(cfg->sdata_gpio), "Can't find ac97-sdata gpio\n"); gpiod_set_consumer_name(cfg->sdata_gpio, "AC97 link sdata"); cfg->reset_gpio = devm_gpiod_get_index(dev, "ac97", 2, GPIOD_ASIS); if (IS_ERR(cfg->reset_gpio)) - return dev_err_probe(dev, PTR_ERR(cfg->sync_gpio), "Can't find ac97-reset gpio\n"); + return dev_err_probe(dev, PTR_ERR(cfg->reset_gpio), "Can't find ac97-reset gpio\n"); gpiod_set_consumer_name(cfg->reset_gpio, "AC97 link reset"); return 0; -- 2.25.1

2 1

[PATCH] ASoC: SMA1303: Change the value for right output
by Kiseok Jo 17 Feb '23

17 Feb '23

This device can output mono, left or right. LR data should be swapped to output right data. Signed-off-by: Kiseok Jo <kiseok.jo(a)irondevice.com> --- sound/soc/codecs/sma1303.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/sma1303.c b/sound/soc/codecs/sma1303.c index 727c01facf52..fa4b0a60f8a9 100644 --- a/sound/soc/codecs/sma1303.c +++ b/sound/soc/codecs/sma1303.c @@ -569,7 +569,7 @@ static int sma1303_aif_in_event(struct snd_soc_dapm_widget *w, ret += sma1303_regmap_update_bits(sma1303, SMA1303_11_SYSTEM_CTRL2, SMA1303_LR_DATA_SW_MASK, - SMA1303_LR_DATA_SW_NORMAL, + SMA1303_LR_DATA_SW_SWAP, &temp); if (temp == true) change = true; base-commit: ab2932214588224b11102e40ba2e78a6f099b8ef -- 2.20.1

1 0

[PATCH] ASoC: Intel: Skylake: Fix struct definition
by Amadeusz Sławiński 16 Feb '23

16 Feb '23

The kernel is globally removing the ambiguous 0-length and 1-element arrays in favor of flexible arrays, so that we can gain both compile-time and run-time array bounds checking[1]. In this instance, struct skl_cpr_cfg contains struct skl_cpr_gtw_cfg, which defined "config_data" as a 1-element array. However, case present in sound/soc/intel/skylake/skl-topology.h is not a simple one as the structure takes part in IPC communication. Apparently original definition missed one field, which while not used by AudioDSP firmware when there is no additional data, is still expected to be part of an IPC message. Currently this works because of how 'config_data' is declared: 'config_data[1]'. Now when one replaces it with a flexible array there would be one field missing. Update struct declaration to fix this. Reported-by: Sasa Ostrouska <casaxa(a)gmail.com> Link: https://lore.kernel.org/all/CALFERdwvq5day_sbDfiUsMSZCQu9HG8-SBpOZDNPeMdZGo… Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Cc: Bard Liao <yung-chuan.liao(a)linux.intel.com> Cc: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Cc: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: "Gustavo A. R. Silva" <gustavoars(a)kernel.org> Cc: alsa-devel(a)alsa-project.org CC: Kees Cook <keescook(a)chromium.org> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/intel/skylake/skl-messages.c | 2 +- sound/soc/intel/skylake/skl-topology.h | 5 ++++- 2 files changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index 5ab0917a2b3d..d31509298a0a 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -549,7 +549,7 @@ static void skl_copy_copier_caps(struct skl_module_cfg *mconfig, if (mconfig->formats_config[SKL_PARAM_INIT].caps_size == 0) return; - memcpy(cpr_mconfig->gtw_cfg.config_data, + memcpy(&cpr_mconfig->gtw_cfg.config_data, mconfig->formats_config[SKL_PARAM_INIT].caps, mconfig->formats_config[SKL_PARAM_INIT].caps_size); diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index 6db0fd7bad49..30a0977af943 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -115,7 +115,10 @@ struct skl_cpr_gtw_cfg { u32 dma_buffer_size; u32 config_length; /* not mandatory; required only for DMIC/I2S */ - u32 config_data[1]; + struct { + u32 gtw_attrs; + u32 data[]; + } config_data; } __packed; struct skl_dma_control { -- 2.34.1

4 3

[PATCH 14/17] dt-bindings: sound: apple,mca: Add t8112-mca compatible
by Janne Grunau 16 Feb '23

16 Feb '23

The block found on Apple's M2 SoC is compatible with the existing driver so add its per-SoC compatible. Signed-off-by: Janne Grunau <j(a)jannau.net> --- This trivial dt-bindings update should be merged through the asahi-soc tree to ensure validation of the Apple M2 (t8112) devicetrees in this series. --- Documentation/devicetree/bindings/sound/apple,mca.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/apple,mca.yaml b/Documentation/devicetree/bindings/sound/apple,mca.yaml index 40e3a202f443..5c6ec08c7d24 100644 --- a/Documentation/devicetree/bindings/sound/apple,mca.yaml +++ b/Documentation/devicetree/bindings/sound/apple,mca.yaml @@ -23,6 +23,7 @@ properties: - enum: - apple,t6000-mca - apple,t8103-mca + - apple,t8112-mca - const: apple,mca reg: -- 2.39.1

3 4

[PATCH 0/9] Refactor Vangogh acp5x machine driver
by Lucas Tanure 16 Feb '23

16 Feb '23

Provide small fixes and refactor the code for easier insertion of a new platform using the same acp5x machine driver. Lucas Tanure (9): ASoC: amd: vangogh: Remove unnecessary init function ASoC: amd: vangogh: Update code indentation ASoC: amd: vangogh: use sizeof of variable instead of struct type ASoC: amd: vangogh: remove unnecessarily included headers ASoC: amd: vangogh: use for_each_rtd_components instead of for ASoC: amd: vangogh: Check Bit Clock rate before snd_soc_dai_set_pll ASoC: amd: vangogh: Move nau8821 and CPU side code up for future platform ASoC: amd: vangogh: Centralize strings definition ASoC: amd: vangogh: Include cs35l41 in structs names sound/soc/amd/vangogh/acp5x-mach.c | 313 +++++++++++++---------------- 1 file changed, 141 insertions(+), 172 deletions(-) -- 2.39.2

2 11

[PATCH] pcm_rate: fix hw_ptr boundary wrapping issue
by Alan Young 16 Feb '23

16 Feb '23

Wrap the hw_ptr using the total position of the slave hw_ptr, including boundary wraps. Otherwise, small errors can creep in due to residuals (when boundary is not a multiple of period size) and which can accumulate. Signed-off-by: Alan Young <consult.awy(a)gmail.com> Fixes: 7570e5d7 ("pcm: rate: fix the hw_ptr update until the boundary available") --- src/pcm/pcm_rate.c | 54 ++++++++++++++++++++++++++++++---------------- 1 file changed, 36 insertions(+), 18 deletions(-) diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index c8076859..a29fc5a9 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -52,6 +52,7 @@ struct _snd_pcm_rate { snd_pcm_generic_t gen; snd_pcm_uframes_t appl_ptr, hw_ptr, last_slave_hw_ptr; snd_pcm_uframes_t last_commit_ptr; + u_int64_t slave_hw_ptr_wrap; snd_pcm_uframes_t orig_avail_min; snd_pcm_sw_params_t sw_params; snd_pcm_format_t sformat; @@ -562,6 +563,8 @@ static int snd_pcm_rate_init(snd_pcm_t *pcm) if (rate->ops.reset) rate->ops.reset(rate->obj); rate->last_commit_ptr = 0; + rate->last_slave_hw_ptr = 0; + rate->slave_hw_ptr_wrap = 0; rate->start_pending = 0; return 0; } @@ -638,33 +641,48 @@ snd_pcm_rate_read_areas1(snd_pcm_t *pcm, static inline void snd_pcm_rate_sync_hwptr0(snd_pcm_t *pcm, snd_pcm_uframes_t slave_hw_ptr) { snd_pcm_rate_t *rate; - snd_pcm_sframes_t slave_hw_ptr_diff; - snd_pcm_sframes_t last_slave_hw_ptr_frac; + snd_pcm_uframes_t new_hw_ptr; + snd_pcm_uframes_t slave_residual; if (pcm->stream != SND_PCM_STREAM_PLAYBACK) return; rate = pcm->private_data; - slave_hw_ptr_diff = pcm_frame_diff(slave_hw_ptr, rate->last_slave_hw_ptr, rate->gen.slave->boundary); - if (slave_hw_ptr_diff == 0) + if (slave_hw_ptr == rate->last_slave_hw_ptr) return; - last_slave_hw_ptr_frac = rate->last_slave_hw_ptr % rate->gen.slave->period_size; - /* While handling fraction part fo slave period, rounded value will be - * introduced by input_frames(). - * To eliminate rounding issue on rate->hw_ptr, subtract last rounded - * value from rate->hw_ptr and add new rounded value of present - * slave_hw_ptr fraction part to rate->hw_ptr. Hence, - * rate->hw_ptr += [ (no. of updated slave periods * pcm rate period size) - - * fractional part of last_slave_hw_ptr rounded value + - * fractional part of updated slave hw ptr's rounded value ] + + /* + * Our hw_ptr may wrap at a different time to that of the slave as the + * number of period / boundary may differ. It is essential that our hw_ptr + * wraps at the correct point, along with our appl_ptr. Otherwise the + * avail calculation will be incorrect. + * + * Track the number of slave hw_ptr boundary wraps and use that to calculate + * our new_hw_ptr from the total number of samples converted. */ - rate->hw_ptr += ( - (((last_slave_hw_ptr_frac + slave_hw_ptr_diff) / rate->gen.slave->period_size) * pcm->period_size) - - rate->ops.input_frames(rate->obj, last_slave_hw_ptr_frac) + - rate->ops.input_frames(rate->obj, (last_slave_hw_ptr_frac + slave_hw_ptr_diff) % rate->gen.slave->period_size)); + if (slave_hw_ptr < rate->last_slave_hw_ptr) { + rate->slave_hw_ptr_wrap += rate->gen.slave->boundary; + } rate->last_slave_hw_ptr = slave_hw_ptr; - rate->hw_ptr %= pcm->boundary; + /* + * Calculate rate->hw_ptr using total number of frames consumed by slave hw_ptr, + * including any boundary wraps. + */ + if (rate->slave_hw_ptr_wrap) { + /* + * Restrict explicit 64-bit calculations to case where rate->slave_hw_ptr_wrap + * is non-zero, which will only happen in 32-bit environments. + */ + u_int64_t wrapped_slave_hw_ptr = slave_hw_ptr + rate->slave_hw_ptr_wrap; + new_hw_ptr = ((wrapped_slave_hw_ptr / rate->gen.slave->period_size) * pcm->period_size) % pcm->boundary; + slave_residual = wrapped_slave_hw_ptr % rate->gen.slave->period_size; + } else { + new_hw_ptr = (slave_hw_ptr / rate->gen.slave->period_size) * pcm->period_size; + slave_residual = slave_hw_ptr % rate->gen.slave->period_size; + } + new_hw_ptr += rate->ops.input_frames(rate->obj, slave_residual); + rate->hw_ptr = new_hw_ptr % pcm->boundary; } static inline void snd_pcm_rate_sync_hwptr(snd_pcm_t *pcm) -- 2.25.1

2 3

[PATCH] ASoC: rsnd: adg: Fix BRG typos
by Geert Uytterhoeven 16 Feb '23

16 Feb '23

"BRG" stands for "Baud Rate Generator", but is frequently misspelled as "RBG". Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- sound/soc/sh/rcar/adg.c | 64 ++++++++++++++++++++--------------------- 1 file changed, 32 insertions(+), 32 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 7bc4421835d7261f..0b8926600d900a68 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -39,10 +39,10 @@ struct rsnd_adg { int clkin_size; int clkout_size; u32 ckr; - u32 rbga; - u32 rbgb; + u32 brga; + u32 brgb; - int rbg_rate[ADG_HZ_SIZE]; /* RBGA / RBGB */ + int brg_rate[ADG_HZ_SIZE]; /* BRGA / BRGB */ }; #define for_each_rsnd_clkin(pos, adg, i) \ @@ -75,7 +75,7 @@ static const char * const clkout_name_gen2[] = { [CLKOUT3] = "audio_clkout3", }; -static u32 rsnd_adg_calculate_rbgx(unsigned long div) +static u32 rsnd_adg_calculate_brgx(unsigned long div) { int i; @@ -131,8 +131,8 @@ static void __rsnd_adg_get_timesel_ratio(struct rsnd_priv *priv, adg->clkin_rate[CLKA], /* 0000: CLKA */ adg->clkin_rate[CLKB], /* 0001: CLKB */ adg->clkin_rate[CLKC], /* 0010: CLKC */ - adg->rbg_rate[ADG_HZ_441], /* 0011: RBGA */ - adg->rbg_rate[ADG_HZ_48], /* 0100: RBGB */ + adg->brg_rate[ADG_HZ_441], /* 0011: BRGA */ + adg->brg_rate[ADG_HZ_48], /* 0100: BRGB */ }; min = ~0; @@ -323,10 +323,10 @@ int rsnd_adg_clk_query(struct rsnd_priv *priv, unsigned int rate) /* * find divided clock from BRGA/BRGB */ - if (rate == adg->rbg_rate[ADG_HZ_441]) + if (rate == adg->brg_rate[ADG_HZ_441]) return 0x10; - if (rate == adg->rbg_rate[ADG_HZ_48]) + if (rate == adg->brg_rate[ADG_HZ_48]) return 0x20; return -EIO; @@ -358,13 +358,13 @@ int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *ssi_mod, unsigned int rate) ckr = 0x80000000; /* BRGB output = 48kHz */ rsnd_mod_bset(adg_mod, BRGCKR, 0x80770000, adg->ckr | ckr); - rsnd_mod_write(adg_mod, BRRA, adg->rbga); - rsnd_mod_write(adg_mod, BRRB, adg->rbgb); + rsnd_mod_write(adg_mod, BRRA, adg->brga); + rsnd_mod_write(adg_mod, BRRB, adg->brgb); dev_dbg(dev, "CLKOUT is based on BRG%c (= %dHz)\n", (ckr) ? 'B' : 'A', - (ckr) ? adg->rbg_rate[ADG_HZ_48] : - adg->rbg_rate[ADG_HZ_441]); + (ckr) ? adg->brg_rate[ADG_HZ_48] : + adg->brg_rate[ADG_HZ_441]); return 0; } @@ -484,7 +484,7 @@ static int rsnd_adg_get_clkout(struct rsnd_priv *priv) struct device *dev = rsnd_priv_to_dev(priv); struct device_node *np = dev->of_node; struct property *prop; - u32 ckr, rbgx, rbga, rbgb; + u32 ckr, brgx, brga, brgb; u32 rate, div; u32 req_rate[ADG_HZ_SIZE] = {}; uint32_t count = 0; @@ -501,8 +501,8 @@ static int rsnd_adg_get_clkout(struct rsnd_priv *priv) }; ckr = 0; - rbga = 2; /* default 1/6 */ - rbgb = 2; /* default 1/6 */ + brga = 2; /* default 1/6 */ + brgb = 2; /* default 1/6 */ /* * ADG supports BRRA/BRRB output only @@ -543,30 +543,30 @@ static int rsnd_adg_get_clkout(struct rsnd_priv *priv) if (0 == rate) /* not used */ continue; - /* RBGA */ - if (!adg->rbg_rate[ADG_HZ_441] && (0 == rate % 44100)) { + /* BRGA */ + if (!adg->brg_rate[ADG_HZ_441] && (0 == rate % 44100)) { div = 6; if (req_Hz[ADG_HZ_441]) div = rate / req_Hz[ADG_HZ_441]; - rbgx = rsnd_adg_calculate_rbgx(div); - if (BRRx_MASK(rbgx) == rbgx) { - rbga = rbgx; - adg->rbg_rate[ADG_HZ_441] = rate / div; + brgx = rsnd_adg_calculate_brgx(div); + if (BRRx_MASK(brgx) == brgx) { + brga = brgx; + adg->brg_rate[ADG_HZ_441] = rate / div; ckr |= brg_table[i] << 20; if (req_Hz[ADG_HZ_441]) parent_clk_name = __clk_get_name(clk); } } - /* RBGB */ - if (!adg->rbg_rate[ADG_HZ_48] && (0 == rate % 48000)) { + /* BRGB */ + if (!adg->brg_rate[ADG_HZ_48] && (0 == rate % 48000)) { div = 6; if (req_Hz[ADG_HZ_48]) div = rate / req_Hz[ADG_HZ_48]; - rbgx = rsnd_adg_calculate_rbgx(div); - if (BRRx_MASK(rbgx) == rbgx) { - rbgb = rbgx; - adg->rbg_rate[ADG_HZ_48] = rate / div; + brgx = rsnd_adg_calculate_brgx(div); + if (BRRx_MASK(brgx) == brgx) { + brgb = brgx; + adg->brg_rate[ADG_HZ_48] = rate / div; ckr |= brg_table[i] << 16; if (req_Hz[ADG_HZ_48]) parent_clk_name = __clk_get_name(clk); @@ -620,8 +620,8 @@ static int rsnd_adg_get_clkout(struct rsnd_priv *priv) rsnd_adg_get_clkout_end: adg->ckr = ckr; - adg->rbga = rbga; - adg->rbgb = rbgb; + adg->brga = brga; + adg->brgb = brgb; return 0; @@ -663,9 +663,9 @@ void rsnd_adg_clk_dbg_info(struct rsnd_priv *priv, struct seq_file *m) __clk_get_name(clk), clk, clk_get_rate(clk)); dbg_msg(dev, m, "BRGCKR = 0x%08x, BRRA/BRRB = 0x%x/0x%x\n", - adg->ckr, adg->rbga, adg->rbgb); - dbg_msg(dev, m, "BRGA (for 44100 base) = %d\n", adg->rbg_rate[ADG_HZ_441]); - dbg_msg(dev, m, "BRGB (for 48000 base) = %d\n", adg->rbg_rate[ADG_HZ_48]); + adg->ckr, adg->brga, adg->brgb); + dbg_msg(dev, m, "BRGA (for 44100 base) = %d\n", adg->brg_rate[ADG_HZ_441]); + dbg_msg(dev, m, "BRGB (for 48000 base) = %d\n", adg->brg_rate[ADG_HZ_48]); /* * Actual CLKOUT will be exchanged in rsnd_adg_ssi_clk_try_start() -- 2.34.1

3 2

[PATCH 1/2] ALSA: pcm: generate default audio timestamp
by Alan Young 16 Feb '23

16 Feb '23

If there is no get_time_info() available or the requested audio timestamp type is AUDIO_TSTAMP_TYPE_DEFAULT then need to set audio_tstamp_report.actual_type = AUDIO_TSTAMP_TYPE_DEFAULT (in the else clause near the start of snd_pcm_update_hw_ptr0()) in order for update_audio_tstamp() to generate the audio timestamp from the number of frames. Fixes: 3179f6200188 ("ALSA: core: add .get_time_info") Signed-off-by: consult.awy(a)gmail.com Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/core/pcm_lib.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 8b6aeb8a78f7..943f5396fc60 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -289,8 +289,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, /* re-test in case tstamp type is not supported in hardware and was demoted to DEFAULT */ if (runtime->audio_tstamp_report.actual_type == SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT) snd_pcm_gettime(runtime, &curr_tstamp); - } else + } else { + runtime->audio_tstamp_report.actual_type = SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT; snd_pcm_gettime(runtime, &curr_tstamp); + } } if (pos == SNDRV_PCM_POS_XRUN) { -- 2.31.1

1 0

[PATCH 2/2] ALSA: Change calculation of audio time from frames
by Alan Young 16 Feb '23

16 Feb '23

Introduce snd_pcm_lib_frames_to_timespec64() to replace the calculation audio_nsecs = div_u64(audio_frames * 1000000000LL, runtime->rate); which overflows after a playback duration of 4.8 days @ 44100 HZ or 0.56 days @ 384000 Hz. The implementation of snd_pcm_lib_frames_to_timespec64() extends these limits, to 1.5 million years (@ 384000 Hz), or the 68 year limit of struct timespec on 32-bit systems. Fixes: 3179f6200188 ("ALSA: core: add .get_time_info") Fixes: 4eeaaeaea1ce ("ALSA: core: add hooks for audio timestamps") Signed-off-by: consult.awy(a)gmail.com Suggested-by: Jaroslav Kysela <perex(a)perex.cz> Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/core/pcm_lib.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 943f5396fc60..e0a6556d33ee 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -205,12 +205,19 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, return 0; } +static inline void snd_pcm_lib_frames_to_timespec64(u64 frames, unsigned int rate, struct timespec64 *audio_tstamp) +{ + u32 remainder; + audio_tstamp->tv_sec = div_u64_rem(frames, rate, &remainder); + audio_tstamp->tv_nsec = div_u64(mul_u32_u32(remainder, NSEC_PER_SEC), rate); +} + static void update_audio_tstamp(struct snd_pcm_substream *substream, struct timespec64 *curr_tstamp, struct timespec64 *audio_tstamp) { struct snd_pcm_runtime *runtime = substream->runtime; - u64 audio_frames, audio_nsecs; + u64 audio_frames; struct timespec64 driver_tstamp; if (runtime->tstamp_mode != SNDRV_PCM_TSTAMP_ENABLE) @@ -233,9 +240,7 @@ static void update_audio_tstamp(struct snd_pcm_substream *substream, else audio_frames += runtime->delay; } - audio_nsecs = div_u64(audio_frames * 1000000000LL, - runtime->rate); - *audio_tstamp = ns_to_timespec64(audio_nsecs); + snd_pcm_lib_frames_to_timespec64(audio_frames, runtime->rate, audio_tstamp); } if (runtime->status->audio_tstamp.tv_sec != audio_tstamp->tv_sec || -- 2.31.1

1 0

[PATCH 1/2] ALSA: pcm: generate default audio timestamp
by Alan Young 16 Feb '23

16 Feb '23

If there is no get_time_info() available or the requested audio timestamp type is AUDIO_TSTAMP_TYPE_DEFAULT then need to set audio_tstamp_report.actual_type = AUDIO_TSTAMP_TYPE_DEFAULT (in the else clause near the start of snd_pcm_update_hw_ptr0()) in order for update_audio_tstamp() to generate the audio timestamp from the number of frames. Fixes: 3179f6200188 ("ALSA: core: add .get_time_info") Signed-off-by: consult.awy(a)gmail.com --- sound/core/pcm_lib.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 8b6aeb8a78f7..943f5396fc60 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -289,8 +289,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, /* re-test in case tstamp type is not supported in hardware and was demoted to DEFAULT */ if (runtime->audio_tstamp_report.actual_type == SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT) snd_pcm_gettime(runtime, &curr_tstamp); - } else + } else { + runtime->audio_tstamp_report.actual_type = SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT; snd_pcm_gettime(runtime, &curr_tstamp); + } } if (pos == SNDRV_PCM_POS_XRUN) { -- 2.31.1

2 1

[PATCH 2/2] ALSA: Change calculation of audio time from frames
by Alan Young 16 Feb '23

16 Feb '23

Introduce snd_pcm_lib_frames_to_timespec64() to replace the calculation audio_nsecs = div_u64(audio_frames * 1000000000LL, runtime->rate); which overflows after a playback duration of 4.8 days @ 44100 HZ or 0.56 days @ 384000 Hz. The implementation of snd_pcm_lib_frames_to_timespec64() extends these limits, to 1.5 million years (@ 384000 Hz), or the 68 year limit of struct timespec on 32-bit systems. Fixes: 3179f6200188 ("ALSA: core: add .get_time_info") Fixes: 4eeaaeaea1ce ("ALSA: core: add hooks for audio timestamps") Signed-off-by: consult.awy(a)gmail.com --- sound/core/pcm_lib.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 943f5396fc60..e0a6556d33ee 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -205,12 +205,19 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, return 0; } +static inline void snd_pcm_lib_frames_to_timespec64(u64 frames, unsigned int rate, struct timespec64 *audio_tstamp) +{ + u32 remainder; + audio_tstamp->tv_sec = div_u64_rem(frames, rate, &remainder); + audio_tstamp->tv_nsec = div_u64(mul_u32_u32(remainder, NSEC_PER_SEC), rate); +} + static void update_audio_tstamp(struct snd_pcm_substream *substream, struct timespec64 *curr_tstamp, struct timespec64 *audio_tstamp) { struct snd_pcm_runtime *runtime = substream->runtime; - u64 audio_frames, audio_nsecs; + u64 audio_frames; struct timespec64 driver_tstamp; if (runtime->tstamp_mode != SNDRV_PCM_TSTAMP_ENABLE) @@ -233,9 +240,7 @@ static void update_audio_tstamp(struct snd_pcm_substream *substream, else audio_frames += runtime->delay; } - audio_nsecs = div_u64(audio_frames * 1000000000LL, - runtime->rate); - *audio_tstamp = ns_to_timespec64(audio_nsecs); + snd_pcm_lib_frames_to_timespec64(audio_frames, runtime->rate, audio_tstamp); } if (runtime->status->audio_tstamp.tv_sec != audio_tstamp->tv_sec || -- 2.31.1

2 1

[GIT PULL] ASoC updates for v6.3
by Mark Brown 16 Feb '23

16 Feb '23

The following changes since commit c1619ea22d9509cfbcf5c2c1cc65563179e9dc8e: ASoC: SOF: sof-audio: Fixes for widget prepare and (2023-01-18 15:28:17 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v6.3 for you to fetch changes up to 76f5aaabce492aa6991c28c96bb78b00b05d06c5: ASoC: soc-ac97: Return correct error codes (2023-02-15 16:09:07 +0000) ---------------------------------------------------------------- ASoC: Updates for v6.3 There's been quite a lot of activity this release, but not really one big feature - lots of new devices, plus a lot of cleanup and modernisation work spread throughout the subsystem: - More factoring out of common operations into helper functions by Morimoto-san. - DT schema conversons and stylistic nits. - Continued work on building out the new SOF IPC4 scheme. - Support for Awinc AT88395, Infineon PEB2466, Iron Device SMA1303, Mediatek MT8188, Realtek RT712, Renesas IDT821034, Samsung/Tesla FSD SoC I2S, and TI TAS5720A-Q1. ---------------------------------------------------------------- Ajye Huang (2): ASoC: mediatek: mt8186-rt5682: primary_codec_init() warn: missing error code? 'ret' ASoC: nau8315: remove dependency on GPIOLIB Alexander Stein (2): ASoC: fsl_sai: Use dev_err_probe ASoC: imx-hdmi: Use dev_err_probe Alexander Sverdlin (1): ASoC: ep93xx: ignore 0 Hz sysclk Alexandru Ardelean (1): ASoC: codecs: tas5720: split a tas5720_mute_soc_component() function Amadeusz Sławiński (18): ASoC: Intel: avs: Add peakvol runtime-parameter requests ASoC: Intel: avs: Add control volume operations ASoC: Intel: avs: Parse control tuples ASoC: Intel: avs: Peakvol module configuration ASoC: Intel: avs: Correctly access topology fields ASoC: Intel: avs: Use min_t instead of min with cast ASoC: topology: Properly access value coming from topology file ASoC: topology: Remove unused SOC_TPLG_PASS_PINS constant ASoC: topology: Fix typo in functions name ASoC: topology: Fix function name ASoC: topology: Rename remove_ handlers ASoC: topology: Remove unnecessary forward declarations ASoC: topology: Pass correct pointer instead of casting ASoC: topology: Return an error on complete() failure ASoC: topology: Remove unnecessary check for EOF ASoC: topology: Unify kcontrol removal code ASoC: topology: Use unload() op directly ASoC: Intel: Skylake: Fix struct definition Andy Shevchenko (7): ASoC: Intel: bytcht_cx2072x: Replace open coded acpi_dev_put() ASoC: Intel: bytcht_da7213: Replace open coded acpi_dev_put() ASoC: Intel: cht_bsw_rt5645: Replace open coded acpi_dev_put() ASoC: Intel: cht_bsw_rt5672: Replace open coded acpi_dev_put() ASoC: Intel: sof-wm8804: Replace open coded acpi_dev_put() ASoC: soc-ac97: Convert to agnostic GPIO API ASoC: soc-ac97: Return correct error codes Arnd Bergmann (4): ASoC: PXA: make SND_PXA2XX_SOC_AC97 user-selectable ASoC: ux500: remove platform_data support ASoC: ux500: remove stedma40 references ASoC: SOF: fix intel-soundwire link failure Astrid Rost (3): ASoC: soc-component: add get_jack_type ASoC: simple-card-utils: create jack inputs for aux_devs ASoC: ts3a227e: add set_jack and get_jack_type Bard Liao (3): ASoC: SOF: ipc4-topology: add buffer type support ASoC: SOF: ipc4-topology: use different channel mask for each sdw amp feedback ASoC: SOF: ipc4-topology: set copier sink format Brent Lu (1): ASoC: Intel: sof_ssp_amp: remove unused variable Cezary Rojewski (2): ASoC: Intel: avs: Use asoc_substream_to_rtd() to obtain rtd ASoC: Intel: avs: Simplify probe-component implementation Chancel Liu (3): ASoC: dt-bindings: fsl,xcvr: Add compatible string for i.MX93 platform ASoC: fsl_xcvr: Add support for i.MX93 platform ASoC: fsl_xcvr: Add constraints of period size while using eDMA Charles Keepax (2): ASoC: cs35l45: Remove separate tables module ASoC: cs35l45: Remove separate namespace for tables Christophe Branchereau (1): ASoC: codecs/jz4760: add digital gain controls Chunxu Li (1): ASoC: mediatek: mt8186: add AFE_DAC_CON0 to volatile register list Claudiu Beznea (12): ASoC: dt-bindings: microchip: use proper naming syntax ASoC: mchp-pdmc: use runtime pm for clock power saving ASoC: mchp-pdmc: add support for suspend to RAM ASoC: mchp-spdiftx: use FIELD_PREP() where possible ASoC: mchp-spdifrx: fix controls which rely on rsr register ASoC: mchp-spdifrx: fix return value in case completion times out ASoC: mchp-spdifrx: fix controls that works with completion mechanism ASoC: mchp-spdifrx: disable all interrupts in mchp_spdifrx_dai_remove() ASoC: mchp-spdifrx: use unsigned long to store clk_get_rate() value ASoC: mchp-spdifrx: remove struct mchp_spdifrx_dev::fmt member ASoC: mchp-spdifrx: add runtime pm support ASoC: mchp-spdifrx: document data structures Colin Ian King (2): ASoC: SMA1303: Fix spelling mistake "Invald" -> "Invalid" ASoC: codecs: aw88395: Fix spelling mistake "cersion" -> "version" Dan Carpenter (1): ASoC: mediatek: mt8188: remove some dead code Daniel Baluta (4): ASoC: SOF: Prepare ipc_msg_data to be used with compress API ASoC: SOF: Prepare set_stream_data_offset for compress API ASoC: SOF: Add support for compress API for stream data/offset ASoC: SOF: compress: Set compress data offset David Rau (1): ASoC: da7213: Add support for mono, set frame width to 32 when possible Deepak R Varma (2): ASoC: imx-pcm-rpmsg: Remove unused variable ASoC: codecs: wcd934x: Use min macro for comparison and assignment Emanuele Ghidoli (3): ASoC: dt-bindings: nau8822: convert to the dtschema ASoC: nau8822: add speaker Bridge Tied Output configuration ASoC: dt-bindings: nau8822: add nuvoton,spk-btl property to dtschema Faiz Abbas (1): ASoC: dt-bindings: simple-card: Document simple-audio-card,plat Geert Uytterhoeven (1): ASoC: dt-bindings: ti,pcm3168a: Convert to json-schema Herve Codina (6): ASoC: Add Renesas IDT821034 codec bindings ASoC: codecs: Add support for the Renesas IDT821034 codec MAINTAINERS: add the Renesas IDT821034 codec entry ASoC: dt-bindings: Add the Infineon PEB2466 codec ASoC: codecs: Add support for the Infineon PEB2466 codec MAINTAINERS: add the Infineon PEB2466 codec entry Jai Luthra (2): dt-bindings: sound: tlv320aic3x: Convert to dtschema ASoC: tlv320aic3x: Add optional clock and port properties Jerome Brunet (9): ASoC: dt-bindings: create component common schema ASoC: dt-bindings: meson: fix gx-card codec node regex ASoC: dt-bindings: meson: convert axg tdm interface to schema ASoC: dt-bindings: meson: convert axg tdm formatters to schema ASoC: dt-bindings: meson: convert axg pdm to schema ASoC: dt-bindings: meson: convert axg fifo to schema ASoC: dt-bindings: meson: convert axg spdif input to schema ASoC: dt-bindings: meson: convert axg spdif output to schema ASoC: dt-bindings: meson: convert axg sound card control to schema Jiapeng Chong (1): ASoC: codecs: Fix unsigned comparison with less than zero Judy Hsiao (1): ASoC: qcom: lpass-platform: Use SNDRV_DMA_TYPE_NONCOHERENT page allocation Kees Cook (1): ASoC: kirkwood: Iterate over array indexes instead of using pointer math Kiseok Jo (7): ASoC: Modified the schema binding and added the vendor prefixes. MAINTAINERS: add IRON DEVICE AUDIO CODEC DRIVERS ASoC: dt-bindings: irondevice,sma1303.yaml: Fix about breaking the checks ASoC: SMA1303: Remove the I2C Retry property in devicetree ASoC: SMA1303: Convert the TDM slot properties in devicetree to mixer ASoC: dt-bindings: irondevice,sma1303: Rework binding and add missing properties ASoC: SMA1303: Remove the sysclk setting in devicetree KiseokJo (1): ASoC: The Iron Device SMA1303 is a boosted Class-D audio amplifier. Krzysztof Kozlowski (28): ASoC: dt-bindings: adi,adau7002: Convert to DT schema ASoC: dt-bindings: cirrus,cs35l41: add interrupts ASoC: dt-bindings: cirrus,cs35l41: cleanup $ref and example ASoC: dt-bindings: qcom,lpass-cpu: Document required-opps ASoC: dt-bindings: qcom,lpass-cpu: Fix DAI children pattern ASoC: dt-bindings: qcom,lpass-cpu: Correct and constrain clocks, interrupts, reg ASoC: dt-bindings: simple-card: correct DAI subnodes pattern ASoC: dt-bindings: qcom,lpass-wsa-macro: correct clocks on SM8250 ASoC: dt-bindings: qcom,lpass-tx-macro: correct clocks on SC7280 ASoC: codecs: wsa883x: Simplify &pdev->dev in probe ASoC: codecs: wsa881x: Simplify &pdev->dev in probe ASoC: codecs: wsa881x: Simplify with dev_err_probe ASoC: codecs: wsa881x: Use proper shutdown GPIO polarity ASoC: codecs: constify static sdw_slave_ops struct ASoC: codecs: constify static sdw_port_config struct ASoC: codecs: wsa88xx: remove unneeded includes ASoC: dt-bindings: samsung,odroid: correct codec DAI phandles ASoC: dt-bindings: samsung-i2s: drop unneeded assigned-clock* ASoC: dt-bindings: qcom,sm8250: use fallback for SDM845 sound cards ASoC: qcom: sdm845: add remark about unneeded compatibles ASoC: dt-bindings: renesas,rsnd: simplify list of compatibles ASoC: dt-bindings: maxim,max98090: Convert to dtschema ASoC: codecs: max98090: simplify snd_soc_dai_driver ASoC: dt-bindings: qcom,wsa881x: Allow sound-name-prefix ASoC: dt-bindings: qcom,wcd934x: Describe slim-ifc-dev ASoC: dt-bindings: qcom,wcd934x: Allow usage as IFD device ASoC: dt-bindings: qcom,q6apm-dai: adjust iommus for SM8550 ADSP ASoC: dt-bindings: maxim,max98095: Convert to dtschema Kuninori Morimoto (55): ASoC: soc-compress.c: fixup private_data on snd_soc_new_compress() ASoC: dt-bindings: audio-graph-port: use definitions for port/endpoint ASoC: dt-bindings: audio-graph-port: add definitions/ports ASoC: dt-bindings: audio-graph-port: add missing mclk-fs ASoC: dt-bindings: audio-graph-port: add clocks on endpoint ASoC: dt-bindings: audio-graph-port: remove prefix ASoC: dt-bindings: ak4613: enable Of-graph (Audio-Graph-Card) style ASoC: dt-bindings: renesas,rsnd: add missing playback/capture ASoC: dt-bindings: renesas,rsnd: tidyup rcar_sound,src ASoC: dt-bindings: renesas,rsnd: #sound-dai-cells is not mandatory ASoC: dt-bindings: simple-card: add missing #address-cells/#size-cells ASoC: soc-dai.h: add missing snd_soc_dai_set_widget() ASoC: soc-dai.h: add snd_soc_dai_dma_data_set/get() for low level ASoC: soc-dai.h: add snd_soc_dai_tdm_mask_set/get() helper ASoC: soc-dai.h: add snd_soc_dai_get/set_widget_playback/capture() helper ASoC: soc.h: add snd_soc_card_is_instantiated() helper ASoC: hdmi-codec: use helper function ASoC: hda: use helper function ASoC: max: use helper function ASoC: rt: use helper function ASoC: cirrus: use helper function ASoC: spear: use helper function ASoC: sdw-mockup: use helper function ASoC: intel: use helper function ASoC: mediatek: use helper function ASoC: meson: use helper function ASoC: rockchip: use helper function ASoC: sof: use helper function ASoC: tegra: use helper function ASoC: ti: use helper function ASoC: soc-core.c: use helper function ASoC: soc-dapm.c: use helper function ASoC: soc-dai.c: use helper function ASoC: soc-pcm.c: use helper function ASoC: soc-topology.c: use helper function ASoC: soc-dai.h: cleanup Playback/Capture data for snd_soc_dai ASoC: cs42l42: use helper function ASoC: rsnd: check whether playback/capture property exists ASoC: rsnd: fixup #endif position ASoC: rsnd: Remove unnecessary rsnd_dbg_dai_call() ASoC: rsnd: indicate necessary error when clock start failed ASoC: rsnd: indicate warning once if it can't handle requested rule ASoC: rsnd: use same debug message format on clkout ASoC: rsnd: remove unnecessary ADG flags ASoC: rsnd: rename clk to clkin ASoC: rsnd: moves clkout_name to top of the file ASoC: rsnd: use clkin/out_size ASoC: rsnd: use array for 44.1kHz/48kHz rate handling ASoC: rsnd: tidyup rsnd_dma_addr() ASoC: rsnd: dma.c: tidyup rsnd_dma_probe() ASoC: dt-bindings: renesas: add R8A779G0 V4H ASoC: rsnd: add R-Car Gen4 Sound support ASoC: dt-bindings: renesas,rsnd.yaml: tidyup reg/reg-name ASoC: rsnd: core.c: indicate warning if strange TDM width was set ASoC: dt-bindings: renesas,rsnd.yaml: drop "dmas/dma-names" from "rcar_sound,ssi" Luca Ceresoli (1): ASoC: fsl-asoc-card: constify fsl_asoc_card_dai Lucas Tanure (1): ASoC: soc-dapm.h: fixup warning struct snd_pcm_substream not declared Lukasz Majewski (3): ASoC: wm8940: Remove warning when no plat data present ASoC: wm8940: Rewrite code to set proper clocks ASoC: wm8940: Read chip ID when wm8940 codec probing Marek Vasut (1): ASoC: dt-bindings: fsl-sai: Simplify the VFxxx dmas binding Mario Limonciello (6): ASoC: amd: yc: Adjust the gain for PDM DMIC ASoC: amd: yc: Add a module parameter to influence pdm_gain ASoC: amd: renoir: Adjust the gain for PDM DMIC ASoC: amd: renoir: Add a module parameter to influence pdm_gain ASoC: amd: ps: Adjust the gain for PDM DMIC ASoC: amd: ps: Add a module parameter to influence pdm_gain Mark Brown (31): Add support for DSP volume controls ASoC: Drop empty platform remove functions ASoC: SOF: Extend the IPC ops optionality ASoC: microchip: power saving features and cleanups Add support for XCVR on i.MX93 platform ASoC: nau8822: add speaker Bridge Tied Output ASoC: fsl-asoc-card: Log error code when we fail to register Add support of MediaTek mt8188 to SOF Add tdm support in acp machine driver Fixes for avs driver ASoC: samsung: fsd: audio support for FSD SoC ASoC: Merge up 6.2 fixes Add support to compress API to ipc_msg_data / ASoC: mediatek: Add support for MT8188 SoC Update lpass dt-bindings ASoC: simple-card-utils: create jack inputs for ASoC: SOF: ipc4: Multi-stream playback and capture support ASoC: dt-bindings: audio-graph-port related update ASoC: topology: Fixes and cleanups Add the Renesas IDT821034 codec support ASoC: codecs: Add Awinic AW88395 audio amplifier ASoC: mchp-spdifrx: add runtime PM support and fixes ASoC: use helper function and cleanup ASoC: cs42l42: Add SoundWire support Fix default DMIC gain on AMD PDM drivers ASoC: (SOF) topology: Regression fixes for next ASoC: SOF: core/ipc4/mtl: Add support for PCM delay Add the Infineon PEB2466 codec support ASoC: SOF: ipc4-topology: Configure copier sink format ASoC: qcom: q6dsp and lpass codec stablity fixes ASoC: SOF: Intel/ipc4: Support for low power playback Matthias Kaehlcke (1): SoC: rt5682s: Disable jack detection interrupt during suspend Nathan Chancellor (2): ASoC: amd: ps: Fix uninitialized ret in create_acp64_platform_devs() ASoC: mchp-spdifrx: Fix uninitialized use of mr in mchp_spdifrx_hw_params() Ondrej Jirman (2): ASoC: rt5640: Allow configuration of LOUT to mono differential mode ASoC: rt5640: Allow to describe how LOUT is wired Padmanabhan Rajanbabu (2): ASoC: dt-bindings: Add FSD I2S controller bindings ASoC: samsung: i2s: add support for FSD I2S Peter Ujfalusi (17): ASoC: SOF: sof-audio: Treat tplg_ops->route_setup() as optional ASoC: SOF: sof-audio: Update documentation for sof_ipc_tplg_ops ASoC: SOF: Add helper macro to be used to get an IPC ops ASoC: SOF: pcm: Extend the optionality of IPC ops to IPC as well ASoC: SOF: control: Extend the optionality of IPC ops to IPC as well ASoC: SOF: sof-audio: Extend the optionality of IPC ops to IPC as well ASoC: SOF: topology: Extend the optionality of IPC ops to IPC as well ASoC: SOF: pm: Extend the optionality of IPC ops to IPC as well ASoC: SOF: sof-priv: Mark fw_tracing ops optional in documentation ASoC: SOF: trace: Use sof_ipc_get_ops() in sof_fw_trace_init ASoC: SOF: trace: No need to check for op pointer in sof_fw_trace_free() ASoC: SOF: core: Print out the value of sof_debug if it is set ASoC: SOF: Avoid double decrementing use_count in sof_widget_setup on error ASoC: SOF: Protect swidget->use_count with mutex for kcontrol access race ASoC: SOF: ipc4-pcm: Do not run the trigger pipelines if no spipe is stored ASoC: SOF: topology: Add missed "else" in sof_connect_dai_widget ASoC: topology: Set correct unload callback for graph type Pierre-Louis Bossart (1): ASoC: amd: ps: use static function Rander Wang (14): ASoC: SOF: add ipc4_fw_reg header file ASoC: SOF: add fw_info_box support ASoC: SOF: add time info structure for ipc4 path ASoC: SOF: ipc4-pcm: allocate time info for pcm delay feature ASoC: SOF: ipc4-pcm: add hw_params ASoC: SOF: add delay function support in sof framework ASoC: SOF: add get_stream_position ops for pcm delay ASoC: SOF: Intel: mtl: add get_stream_position support ASoC: SOF: ipc4-pcm: add delay function support ASoC: SOF: Introduce a new set_pm_gate() IPC PM op ASoC: SOF: Intel: hda-dsp: use set_pm_gate according to ipc version ASoC: SOF: ipc4: Wake up dsp core before sending ipc msg ASoC: SOF: Intel: Enable d0i3 work for ipc4 ASoC: SOF: Intel: hda-dsp: Set streaming flag for d0i3 Randy Dunlap (1): ASoC: Kconfig: fix spelling of "up to" Ranjani Sridharan (16): ASoC: SOF: ipc4-topology: No need to unbind routes within a pipeline ASoC: soc-pcm: Export widget_in_list() ASoC: SOF: sof-audio: Set up/free DAI/AIF widgets only once ASoC: SOF: sof-audio: Only process widgets in the connected widget list ASoC: SOF: pcm: do not free widgets during suspend trigger ASoC: SOF: topology: Set IPC-specific trigger order for DAI links ASoC: SOF: Introduce PCM setup/free PCM IPC ops ASoC: SOF: ipc4-pcm: Define pcm_setup/free ops ASoC: SOF: ipc4: Add flag to skip triggering pipelines during FE DAI trigger ASoC: SOF: sof-audio: Populate the PCM stream pipeline_info ASoC: SOF: ipc4-pcm: Use the PCM stream's pipeline_info during trigger ASoC: SOF: Introduce struct snd_sof_pipeline ASoC: SOF: ipc4-pcm: Rename 'data' variable to trigger_list ASoC: SOF: ipc4-pcm: Implement pipeline trigger reference counting ASoC: SOF: ipc4-topology: Protect pipeline free with mutex ASoC: SOF: ipc4-topology: Print queue IDs in error Richard Fitzgerald (6): ASoC: cs42l42: Add SOFT_RESET_REBOOT register ASoC: cs42l42: Ensure MCLKint is a multiple of the sample rate ASoC: cs42l42: Separate ASP config from PLL config ASoC: cs42l42: Export some functions for SoundWire ASoC: cs42l42: Add SoundWire support ASoC: cs42l42: Don't set idle_bias_on Sameer Pujar (1): ASoC: rt5640: Update MCLK rate in set_sysclk() Seppo Ingalsuo (1): ASoC: SOF: IPC3 topology: Print the conflicting bytes sizes Shengjiu Wang (1): ASoC: fsl_sai: initialize is_dsp_mode flag Shuming Fan (3): ASoC: rt711-sdca: add jack detection mode for JD2 100K ASoC: rt712-sdca: Add RT712 SDCA driver for Jack and Amp topology ASoC: rt712-sdca: fix coding style and unconditionally return issues Srinivas Kandagatla (8): ASoC: qcom: q6apm-lpass-dai: unprepare stream if its already prepared ASoC: qcom: q6apm-dai: fix race condition while updating the position pointer ASoC: qcom: q6apm-dai: Add SNDRV_PCM_INFO_BATCH flag ASoC: qcom: audioreach: fix ADSP ready check ASoC: codecs: lpass: register mclk after runtime pm ASoC: codecs: lpass: fix incorrect mclk rate ASoC: codecs: lpass: do not reset soundwire block on clk enable ASoC: codecs: lpass: remove not so useful verbose log Srinivasa Rao Mandadapu (3): ASoC: qdsp6: audioreach: Add ADSP ready check ASoC: qcom: dt-bindings: lpass-va-macro: Update clock name ASoC: dt-bindings: google,sc7280-herobrine: Add platform property Stefan Binding (2): soundwire: stream: Add specific prep/deprep commands to port_prep callback ASoC: cs42l42: Wait for debounce interval after resume Steffen Aschbacher (5): ASoC: codecs: tas5720: add support for TAS5720A-Q1 (automotive) variant ASoC: tas5720: set bit 7 in ANALOG_CTRL_REG for TAS5720A-Q1 during probe ASoC: dt-bindings: add entry for TAS5720A-Q1 driver ASoC: tlv320adcx140: fix 'ti,gpio-config' DT property init ASoC: tlv320adcx140: extend list of supported samplerates Tinghan Shen (3): dt-bindings: dsp: mediatek: Add mt8188 dsp compatible ASoC: SOF: mediatek: Support mt8188 platform ASoC: SOF: mediatek: Provide debugfs_add_region_item ops for core Tom Rix (1): ASoC: codecs: aw88395: initialize cur_scene_id to 0 Trevor Wu (14): ASoC: mediatek: common: add SMC ops and SMC CMD ASoC: mediatek: mt8188: add common header ASoC: mediatek: mt8188: support audsys clock ASoC: mediatek: mt8188: support adda in platform driver ASoC: mediatek: mt8188: support etdm in platform driver ASoC: mediatek: mt8188: support pcmif in platform driver ASoC: mediatek: mt8188: support audio clock control ASoC: mediatek: mt8188: add platform driver ASoC: mediatek: mt8188: add control for timing select ASoC: dt-bindings: mediatek,mt8188-afe: add audio afe document ASoC: mediatek: common: add soundcard driver common code ASoC: mediatek: mt8188: add machine driver with mt6359 ASoC: dt-bindings: mediatek,mt8188-mt6359: add mt8188-mt6359 document ASoC: mediatek: mt8188: remove etdm dead code Uwe Kleine-König (17): ASoC: pxa: Drop empty platform remove function ASoC: amd: Drop empty platform remove function ASoC: atmel-classd: Drop empty platform remove function ASoC: atmel-pdmic: Drop empty platform remove function ASoC: 88pm860x: Drop empty platform remove function ASoC: ac97: Drop empty platform remove function ASoC: adau7002: Drop empty platform remove function ASoC: bt-sco: Drop empty platform remove function ASoC: cq93vc: Drop empty platform remove function ASoC: mc13783: Drop empty platform remove function ASoC: wl1273: Drop empty platform remove function ASoC: pxa: e740_wm9705: Drop empty platform remove function ASoC: pxa: e750_wm9705: Drop empty platform remove function ASoC: pxa: e800_wm9712: Drop empty platform remove function ASoC: SOF: nocodec: Drop empty platform remove function ASoC: pxa: Drop empty platform remove function ASoC: sma1303: Convert to i2c's .probe_new() V sujith kumar Reddy (1): ASoC: SOF: amd: Enable cache for AMD Rembrandt platform Venkata Prasad Potturu (6): ASoC: amd: acp: Add new cpu dai's in machine driver ASoC: amd: acp: Refactor dai format implementation ASoC: amd: acp: Refactor i2s clocks programming sequence ASoC: amd: acp: Add i2s tdm support in machine driver ASoC: amd: acp: Enable i2s tdm support for skyrim platforms ASoC: amd: acp: Refactor bit width calculation Vijendar Mukunda (8): ASoC: amd: ps: implement api to retrieve acp device config ASoC: amd: ps: refactor platform device creation logic ASoC: amd: ps: update dev index value in irq handler ASoC: amd: ps: move irq handler registration ASoC: amd: ps: add mutex lock for accessing common registers ASoC: amd: ps: use acp_lock to protect common registers in pdm driver ASoC: amd: ps: remove unused variable ASoC: amd: update ps platform acp header file Weidong Wang (8): ASoC: codecs: Add i2c and codec registration for aw88395 and their associated operation functions ASoC: codecs: ACF bin parsing and check library file for aw88395 ASoC: codecs: Aw88395 function for ALSA Audio Driver ASoC: codecs: Aw88395 chip register file, data type file and Kconfig Makefile ASoC: dt-bindings: Add schema for "awinic,aw88395" ASoC: codecs: Fixed a spelling error in the function name ASoC: codecs: Modify the log print statement ASoC: codecs: Modify error implicit declaration of function 'gpiod_set_value_cansleep' YC Hung (1): ASoC: SOF: mediatek: mt8195: remove a redundant comparison of sram Yang Li (2): ASoC: codecs: fix platform_no_drv_owner.cocci warning ASoC: codecs: Remove unneeded semicolon Yassine Oudjana (1): ASoC: qcom: apq8096: set driver name correctly 강신형 (1): ASoC: soc-compress: Reposition and add pcm_mutex .../bindings/dsp/mediatek,mt8186-dsp.yaml | 4 +- .../devicetree/bindings/sound/adi,adau7002.txt | 19 - .../devicetree/bindings/sound/adi,adau7002.yaml | 40 + .../devicetree/bindings/sound/ak4613.yaml | 7 + .../devicetree/bindings/sound/amlogic,axg-fifo.txt | 34 - .../bindings/sound/amlogic,axg-fifo.yaml | 112 + .../devicetree/bindings/sound/amlogic,axg-pdm.txt | 29 - .../devicetree/bindings/sound/amlogic,axg-pdm.yaml | 82 + .../bindings/sound/amlogic,axg-sound-card.txt | 124 - .../bindings/sound/amlogic,axg-sound-card.yaml | 183 ++ .../bindings/sound/amlogic,axg-spdifin.txt | 27 - .../bindings/sound/amlogic,axg-spdifin.yaml | 86 + .../bindings/sound/amlogic,axg-spdifout.txt | 25 - .../bindings/sound/amlogic,axg-spdifout.yaml | 79 + .../bindings/sound/amlogic,axg-tdm-formatters.txt | 36 - .../bindings/sound/amlogic,axg-tdm-formatters.yaml | 88 + .../bindings/sound/amlogic,axg-tdm-iface.txt | 22 - .../bindings/sound/amlogic,axg-tdm-iface.yaml | 55 + .../bindings/sound/amlogic,gx-sound-card.yaml | 2 +- .../bindings/sound/audio-graph-port.yaml | 65 +- .../devicetree/bindings/sound/awinic,aw88395.yaml | 53 + .../devicetree/bindings/sound/cirrus,cs35l41.yaml | 43 +- .../bindings/sound/component-common.yaml | 21 + .../devicetree/bindings/sound/dai-common.yaml | 11 +- .../devicetree/bindings/sound/fsl,sai.yaml | 38 +- .../devicetree/bindings/sound/fsl,xcvr.yaml | 1 + .../bindings/sound/google,sc7280-herobrine.yaml | 12 + .../bindings/sound/infineon,peb2466.yaml | 91 + .../bindings/sound/irondevice,sma1303.yaml | 48 + .../devicetree/bindings/sound/max98090.txt | 59 - .../devicetree/bindings/sound/max98095.txt | 22 - .../devicetree/bindings/sound/maxim,max98090.yaml | 84 + .../devicetree/bindings/sound/maxim,max98095.yaml | 54 + .../bindings/sound/mediatek,mt8188-afe.yaml | 208 ++ .../bindings/sound/mediatek,mt8188-mt6359.yaml | 97 + ...,i2s-mcc.yaml => microchip,sama7g5-i2smcc.yaml} | 2 +- ...ochip,pdmc.yaml => microchip,sama7g5-pdmc.yaml} | 2 +- ...spdifrx.yaml => microchip,sama7g5-spdifrx.yaml} | 2 +- ...spdiftx.yaml => microchip,sama7g5-spdiftx.yaml} | 2 +- .../devicetree/bindings/sound/nau8822.txt | 16 - .../devicetree/bindings/sound/nuvoton,nau8822.yaml | 46 + .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 136 +- .../bindings/sound/qcom,lpass-va-macro.yaml | 4 +- .../devicetree/bindings/sound/qcom,q6apm-dai.yaml | 3 +- .../devicetree/bindings/sound/qcom,sm8250.yaml | 24 +- .../devicetree/bindings/sound/qcom,wcd934x.yaml | 62 +- .../devicetree/bindings/sound/qcom,wsa881x.yaml | 5 +- .../bindings/sound/renesas,idt821034.yaml | 75 + .../devicetree/bindings/sound/renesas,rsnd.yaml | 59 +- Documentation/devicetree/bindings/sound/rt5640.txt | 3 + .../devicetree/bindings/sound/samsung,odroid.yaml | 3 +- .../devicetree/bindings/sound/samsung-i2s.yaml | 11 +- .../devicetree/bindings/sound/simple-card.yaml | 18 +- .../devicetree/bindings/sound/tas5720.txt | 2 + .../devicetree/bindings/sound/ti,pcm3168a.txt | 56 - .../devicetree/bindings/sound/ti,pcm3168a.yaml | 107 + .../devicetree/bindings/sound/ti,tlv320aic3x.yaml | 165 + .../devicetree/bindings/sound/tlv320aic3x.txt | 97 - .../devicetree/bindings/vendor-prefixes.yaml | 2 + MAINTAINERS | 21 + drivers/soundwire/stream.c | 4 +- include/linux/platform_data/asoc-ux500-msp.h | 20 - include/linux/soundwire/sdw.h | 8 +- include/sound/acp63_chip_offset_byte.h | 751 ++--- include/sound/cs42l42.h | 5 + include/sound/simple_card_utils.h | 3 + include/sound/soc-component.h | 2 + include/sound/soc-dai.h | 90 +- include/sound/soc-dapm.h | 1 + include/sound/soc-dpcm.h | 2 + include/sound/soc-topology.h | 2 +- include/sound/soc.h | 6 + include/sound/sof/ipc4/header.h | 3 + include/uapi/sound/intel/avs/tokens.h | 4 + sound/soc/amd/acp/acp-legacy-mach.c | 13 + sound/soc/amd/acp/acp-mach-common.c | 465 ++- sound/soc/amd/acp/acp-mach.h | 4 + sound/soc/amd/acp/acp-sof-mach.c | 14 + sound/soc/amd/ps/acp63.h | 14 +- sound/soc/amd/ps/pci-ps.c | 211 +- sound/soc/amd/ps/ps-pdm-dma.c | 40 +- sound/soc/amd/raven/acp3x-i2s.c | 8 - sound/soc/amd/renoir/acp3x-pdm-dma.c | 8 +- sound/soc/amd/renoir/rn_acp3x.h | 2 +- sound/soc/amd/yc/acp6x-pdm-dma.c | 8 +- sound/soc/amd/yc/acp6x.h | 2 +- sound/soc/atmel/atmel-classd.c | 6 - sound/soc/atmel/atmel-pdmic.c | 6 - sound/soc/atmel/mchp-pdmc.c | 145 +- sound/soc/atmel/mchp-spdifrx.c | 554 +++- sound/soc/atmel/mchp-spdiftx.c | 8 +- sound/soc/cirrus/ep93xx-ac97.c | 4 +- sound/soc/cirrus/ep93xx-i2s.c | 6 +- sound/soc/codecs/88pm860x-codec.c | 6 - sound/soc/codecs/Kconfig | 69 +- sound/soc/codecs/Makefile | 19 +- sound/soc/codecs/ac97.c | 6 - sound/soc/codecs/adau7002.c | 6 - sound/soc/codecs/aw88395/aw88395.c | 579 ++++ sound/soc/codecs/aw88395/aw88395.h | 58 + sound/soc/codecs/aw88395/aw88395_data_type.h | 142 + sound/soc/codecs/aw88395/aw88395_device.c | 1748 ++++++++++ sound/soc/codecs/aw88395/aw88395_device.h | 194 ++ sound/soc/codecs/aw88395/aw88395_lib.c | 1066 +++++++ sound/soc/codecs/aw88395/aw88395_lib.h | 92 + sound/soc/codecs/aw88395/aw88395_reg.h | 383 +++ sound/soc/codecs/bt-sco.c | 6 - sound/soc/codecs/cq93vc.c | 6 - sound/soc/codecs/cs35l45-i2c.c | 1 - sound/soc/codecs/cs35l45-spi.c | 1 - sound/soc/codecs/cs35l45-tables.c | 12 +- sound/soc/codecs/cs35l45.c | 1 - sound/soc/codecs/cs42l42-sdw.c | 607 ++++ sound/soc/codecs/cs42l42.c | 133 +- sound/soc/codecs/cs42l42.h | 9 +- sound/soc/codecs/da7213.c | 30 +- sound/soc/codecs/da7213.h | 3 + sound/soc/codecs/hda.c | 7 +- sound/soc/codecs/hdmi-codec.c | 11 +- sound/soc/codecs/idt821034.c | 1178 +++++++ sound/soc/codecs/jz4760.c | 9 + sound/soc/codecs/lpass-rx-macro.c | 31 +- sound/soc/codecs/lpass-tx-macro.c | 25 +- sound/soc/codecs/lpass-va-macro.c | 43 +- sound/soc/codecs/lpass-wsa-macro.c | 27 +- sound/soc/codecs/max98090.c | 8 +- sound/soc/codecs/max98373-sdw.c | 5 +- sound/soc/codecs/mc13783.c | 6 - sound/soc/codecs/nau8822.c | 9 + sound/soc/codecs/nau8822.h | 9 + sound/soc/codecs/peb2466.c | 2071 ++++++++++++ sound/soc/codecs/rt1308-sdw.c | 5 +- sound/soc/codecs/rt1316-sdw.c | 7 +- sound/soc/codecs/rt1318-sdw.c | 7 +- sound/soc/codecs/rt5640.c | 9 + sound/soc/codecs/rt5640.h | 2 + sound/soc/codecs/rt5645.c | 2 +- sound/soc/codecs/rt5665.c | 4 +- sound/soc/codecs/rt5668.c | 4 +- sound/soc/codecs/rt5682-sdw.c | 5 +- sound/soc/codecs/rt5682.c | 4 +- sound/soc/codecs/rt5682s.c | 14 +- sound/soc/codecs/rt5682s.h | 1 + sound/soc/codecs/rt700.c | 7 +- sound/soc/codecs/rt711-sdca-sdw.c | 2 +- sound/soc/codecs/rt711-sdca.c | 22 +- sound/soc/codecs/rt711-sdca.h | 8 +- sound/soc/codecs/rt711.c | 7 +- sound/soc/codecs/rt712-sdca-sdw.c | 485 +++ sound/soc/codecs/rt712-sdca-sdw.h | 108 + sound/soc/codecs/rt712-sdca.c | 1340 ++++++++ sound/soc/codecs/rt712-sdca.h | 220 ++ sound/soc/codecs/rt715-sdca-sdw.c | 2 +- sound/soc/codecs/rt715-sdca.c | 5 +- sound/soc/codecs/rt715.c | 5 +- sound/soc/codecs/sdw-mockup.c | 5 +- sound/soc/codecs/sma1303.c | 1820 +++++++++++ sound/soc/codecs/sma1303.h | 609 ++++ sound/soc/codecs/tas5720.c | 128 +- sound/soc/codecs/tas5720.h | 16 +- sound/soc/codecs/tlv320adcx140.c | 2 +- sound/soc/codecs/tlv320adcx140.h | 4 +- sound/soc/codecs/ts3a227e.c | 20 +- sound/soc/codecs/wcd934x.c | 10 +- sound/soc/codecs/wcd938x-sdw.c | 2 +- sound/soc/codecs/wl1273.c | 6 - sound/soc/codecs/wm8940.c | 116 +- sound/soc/codecs/wm8940.h | 3 + sound/soc/codecs/wsa881x.c | 63 +- sound/soc/codecs/wsa883x.c | 22 +- sound/soc/fsl/fsl-asoc-card.c | 4 +- sound/soc/fsl/fsl_sai.c | 6 +- sound/soc/fsl/fsl_xcvr.c | 155 +- sound/soc/fsl/fsl_xcvr.h | 7 + sound/soc/fsl/imx-hdmi.c | 2 +- sound/soc/fsl/imx-pcm-rpmsg.c | 6 +- sound/soc/generic/simple-card-utils.c | 49 + sound/soc/generic/simple-card.c | 4 + sound/soc/intel/atom/sst-atom-controls.c | 9 +- sound/soc/intel/avs/Makefile | 2 +- sound/soc/intel/avs/boards/nau8825.c | 5 +- sound/soc/intel/avs/boards/rt286.c | 2 +- sound/soc/intel/avs/boards/rt298.c | 2 +- sound/soc/intel/avs/control.c | 105 + sound/soc/intel/avs/control.h | 23 + sound/soc/intel/avs/messages.c | 29 + sound/soc/intel/avs/messages.h | 33 + sound/soc/intel/avs/path.c | 62 + sound/soc/intel/avs/pcm.c | 45 +- sound/soc/intel/avs/probes.c | 23 - sound/soc/intel/avs/topology.c | 76 + sound/soc/intel/avs/topology.h | 3 + sound/soc/intel/avs/trace.c | 2 +- sound/soc/intel/boards/bytcht_cx2072x.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/boards/sof_ssp_amp.c | 4 - sound/soc/intel/boards/sof_wm8804.c | 2 +- sound/soc/intel/skylake/skl-messages.c | 2 +- sound/soc/intel/skylake/skl-topology.c | 19 +- sound/soc/intel/skylake/skl-topology.h | 5 +- sound/soc/kirkwood/kirkwood-dma.c | 2 +- sound/soc/mediatek/Kconfig | 23 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/common/Makefile | 2 +- sound/soc/mediatek/common/mtk-base-afe.h | 19 + sound/soc/mediatek/common/mtk-dsp-sof-common.c | 18 +- sound/soc/mediatek/common/mtk-soundcard-driver.c | 79 + sound/soc/mediatek/common/mtk-soundcard-driver.h | 14 + sound/soc/mediatek/mt6797/mt6797-dai-pcm.c | 8 +- sound/soc/mediatek/mt8183/mt8183-dai-pcm.c | 7 +- sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 1 + sound/soc/mediatek/mt8186/mt8186-dai-pcm.c | 7 +- .../mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c | 2 +- sound/soc/mediatek/mt8188/Makefile | 15 + sound/soc/mediatek/mt8188/mt8188-afe-clk.c | 658 ++++ sound/soc/mediatek/mt8188/mt8188-afe-clk.h | 115 + sound/soc/mediatek/mt8188/mt8188-afe-common.h | 151 + sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 3358 ++++++++++++++++++++ sound/soc/mediatek/mt8188/mt8188-audsys-clk.c | 205 ++ sound/soc/mediatek/mt8188/mt8188-audsys-clk.h | 15 + sound/soc/mediatek/mt8188/mt8188-audsys-clkid.h | 83 + sound/soc/mediatek/mt8188/mt8188-dai-adda.c | 632 ++++ sound/soc/mediatek/mt8188/mt8188-dai-etdm.c | 2573 +++++++++++++++ sound/soc/mediatek/mt8188/mt8188-dai-pcm.c | 368 +++ sound/soc/mediatek/mt8188/mt8188-mt6359.c | 785 +++++ sound/soc/mediatek/mt8188/mt8188-reg.h | 3180 ++++++++++++++++++ sound/soc/mediatek/mt8192/mt8192-dai-pcm.c | 8 +- sound/soc/mediatek/mt8195/mt8195-dai-pcm.c | 7 +- sound/soc/meson/aiu-fifo-i2s.c | 4 +- sound/soc/meson/aiu-fifo-spdif.c | 2 +- sound/soc/meson/aiu-fifo.c | 21 +- sound/soc/meson/axg-tdm-interface.c | 34 +- sound/soc/meson/axg-tdmin.c | 2 +- sound/soc/meson/axg-tdmout.c | 2 +- sound/soc/meson/meson-codec-glue.c | 8 +- sound/soc/pxa/Kconfig | 5 +- sound/soc/pxa/e740_wm9705.c | 6 - sound/soc/pxa/e750_wm9705.c | 6 - sound/soc/pxa/e800_wm9712.c | 6 - sound/soc/pxa/spitz.c | 6 - sound/soc/qcom/apq8096.c | 1 + sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/qcom/qdsp6/q6apm-dai.c | 22 +- sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 5 + sound/soc/qcom/qdsp6/q6apm.c | 14 + sound/soc/qcom/qdsp6/q6apm.h | 2 + sound/soc/qcom/qdsp6/q6prm.c | 4 + sound/soc/qcom/sdm845.c | 1 + sound/soc/rockchip/Kconfig | 2 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 4 +- sound/soc/rockchip/rockchip_pdm.c | 2 +- sound/soc/rockchip/rockchip_spdif.c | 2 +- sound/soc/samsung/i2s-regs.h | 1 + sound/soc/samsung/i2s.c | 53 + sound/soc/sh/rcar/adg.c | 169 +- sound/soc/sh/rcar/core.c | 73 +- sound/soc/sh/rcar/dma.c | 57 +- sound/soc/sh/rcar/gen.c | 70 + sound/soc/sh/rcar/rsnd.h | 23 +- sound/soc/sh/rcar/ssi.c | 13 +- sound/soc/sh/rcar/ssiu.c | 15 +- sound/soc/soc-ac97.c | 68 +- sound/soc/soc-component.c | 20 + sound/soc/soc-compress.c | 11 +- sound/soc/soc-core.c | 8 +- sound/soc/soc-dai.c | 13 +- sound/soc/soc-dapm.c | 26 +- sound/soc/soc-pcm.c | 15 +- sound/soc/soc-topology.c | 187 +- sound/soc/sof/amd/acp-dsp-offset.h | 4 + sound/soc/sof/amd/acp-ipc.c | 8 +- sound/soc/sof/amd/acp-loader.c | 7 + sound/soc/sof/amd/acp.h | 7 +- sound/soc/sof/compress.c | 9 + sound/soc/sof/control.c | 44 +- sound/soc/sof/core.c | 4 + sound/soc/sof/intel/Kconfig | 4 +- sound/soc/sof/intel/cnl.c | 2 + sound/soc/sof/intel/hda-dai.c | 92 +- sound/soc/sof/intel/hda-dsp.c | 46 +- sound/soc/sof/intel/hda-ipc.c | 36 +- sound/soc/sof/intel/hda.h | 6 +- sound/soc/sof/intel/mtl.c | 16 + sound/soc/sof/intel/mtl.h | 6 + sound/soc/sof/ipc3-control.c | 46 +- sound/soc/sof/ipc3-pcm.c | 3 +- sound/soc/sof/ipc3-topology.c | 39 +- sound/soc/sof/ipc3.c | 22 +- sound/soc/sof/ipc4-control.c | 33 +- sound/soc/sof/ipc4-fw-reg.h | 155 + sound/soc/sof/ipc4-pcm.c | 573 +++- sound/soc/sof/ipc4-priv.h | 17 + sound/soc/sof/ipc4-topology.c | 212 +- sound/soc/sof/ipc4-topology.h | 60 + sound/soc/sof/ipc4.c | 29 + sound/soc/sof/mediatek/mt8186/mt8186.c | 22 +- sound/soc/sof/mediatek/mt8186/mt8186.h | 10 +- sound/soc/sof/mediatek/mt8195/mt8195.c | 8 +- sound/soc/sof/mediatek/mt8195/mt8195.h | 2 - sound/soc/sof/nocodec.c | 6 - sound/soc/sof/ops.h | 18 +- sound/soc/sof/pcm.c | 36 +- sound/soc/sof/pm.c | 12 +- sound/soc/sof/sof-audio.c | 281 +- sound/soc/sof/sof-audio.h | 67 +- sound/soc/sof/sof-priv.h | 30 +- sound/soc/sof/stream-ipc.c | 53 +- sound/soc/sof/topology.c | 309 +- sound/soc/sof/trace.c | 8 +- sound/soc/spear/spdif_out.c | 3 +- sound/soc/tegra/Kconfig | 4 +- sound/soc/tegra/tegra20_ac97.c | 4 +- sound/soc/tegra/tegra20_i2s.c | 4 +- sound/soc/tegra/tegra20_spdif.c | 3 +- sound/soc/tegra/tegra210_admaif.c | 4 +- sound/soc/tegra/tegra30_i2s.c | 4 +- sound/soc/ti/davinci-i2s.c | 5 +- sound/soc/ti/davinci-mcasp.c | 5 +- sound/soc/ti/davinci-vcif.c | 5 +- sound/soc/ux500/mop500.c | 8 +- sound/soc/ux500/ux500_msp_dai.c | 37 +- sound/soc/ux500/ux500_msp_i2s.c | 66 +- sound/soc/ux500/ux500_msp_i2s.h | 14 +- sound/soc/ux500/ux500_pcm.c | 83 +- 326 files changed, 32241 insertions(+), 3385 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/adi,adau7002.txt create mode 100644 Documentation/devicetree/bindings/sound/adi,adau7002.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-fifo.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.yaml create mode 100644 Documentation/devicetree/bindings/sound/awinic,aw88395.yaml create mode 100644 Documentation/devicetree/bindings/sound/component-common.yaml create mode 100644 Documentation/devicetree/bindings/sound/infineon,peb2466.yaml create mode 100644 Documentation/devicetree/bindings/sound/irondevice,sma1303.yaml delete mode 100644 Documentation/devicetree/bindings/sound/max98090.txt delete mode 100644 Documentation/devicetree/bindings/sound/max98095.txt create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98090.yaml create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98095.yaml create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt8188-afe.yaml create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml rename Documentation/devicetree/bindings/sound/{mchp,i2s-mcc.yaml => microchip,sama7g5-i2smcc.yaml} (97%) rename Documentation/devicetree/bindings/sound/{microchip,pdmc.yaml => microchip,sama7g5-pdmc.yaml} (97%) rename Documentation/devicetree/bindings/sound/{mchp,spdifrx.yaml => microchip,sama7g5-spdifrx.yaml} (95%) rename Documentation/devicetree/bindings/sound/{mchp,spdiftx.yaml => microchip,sama7g5-spdiftx.yaml} (95%) delete mode 100644 Documentation/devicetree/bindings/sound/nau8822.txt create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8822.yaml create mode 100644 Documentation/devicetree/bindings/sound/renesas,idt821034.yaml delete mode 100644 Documentation/devicetree/bindings/sound/ti,pcm3168a.txt create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml delete mode 100644 Documentation/devicetree/bindings/sound/tlv320aic3x.txt delete mode 100644 include/linux/platform_data/asoc-ux500-msp.h create mode 100644 sound/soc/codecs/aw88395/aw88395.c create mode 100644 sound/soc/codecs/aw88395/aw88395.h create mode 100644 sound/soc/codecs/aw88395/aw88395_data_type.h create mode 100644 sound/soc/codecs/aw88395/aw88395_device.c create mode 100644 sound/soc/codecs/aw88395/aw88395_device.h create mode 100644 sound/soc/codecs/aw88395/aw88395_lib.c create mode 100644 sound/soc/codecs/aw88395/aw88395_lib.h create mode 100644 sound/soc/codecs/aw88395/aw88395_reg.h create mode 100644 sound/soc/codecs/cs42l42-sdw.c create mode 100644 sound/soc/codecs/idt821034.c create mode 100644 sound/soc/codecs/peb2466.c create mode 100644 sound/soc/codecs/rt712-sdca-sdw.c create mode 100644 sound/soc/codecs/rt712-sdca-sdw.h create mode 100644 sound/soc/codecs/rt712-sdca.c create mode 100644 sound/soc/codecs/rt712-sdca.h create mode 100644 sound/soc/codecs/sma1303.c create mode 100644 sound/soc/codecs/sma1303.h create mode 100644 sound/soc/intel/avs/control.c create mode 100644 sound/soc/intel/avs/control.h create mode 100644 sound/soc/mediatek/common/mtk-soundcard-driver.c create mode 100644 sound/soc/mediatek/common/mtk-soundcard-driver.h create mode 100644 sound/soc/mediatek/mt8188/Makefile create mode 100644 sound/soc/mediatek/mt8188/mt8188-afe-clk.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-afe-clk.h create mode 100644 sound/soc/mediatek/mt8188/mt8188-afe-common.h create mode 100644 sound/soc/mediatek/mt8188/mt8188-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-audsys-clk.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-audsys-clk.h create mode 100644 sound/soc/mediatek/mt8188/mt8188-audsys-clkid.h create mode 100644 sound/soc/mediatek/mt8188/mt8188-dai-adda.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-dai-etdm.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-mt6359.c create mode 100644 sound/soc/mediatek/mt8188/mt8188-reg.h create mode 100644 sound/soc/sof/ipc4-fw-reg.h

2 1

[GIT PULL] ASoC fixes for v6.2-rc8
by Mark Brown 16 Feb '23

16 Feb '23

The following changes since commit c173ee5b2fa6195066674d66d1d7e191010fb1ff: ASoC: topology: Return -ENOMEM on memory allocation failure (2023-02-07 14:06:26 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.2-rc8 for you to fetch changes up to 5afc7eefe41645259da84898fc55f6f46cb4de47: ASoC: SOF: ops: refine parameters order in function snd_sof_dsp_update8 (2023-02-08 11:51:45 +0000) ---------------------------------------------------------------- ASoC: Fix for v6.2 One non-urgent fix for v6.2, this could possibly wait till the merge window. ---------------------------------------------------------------- Rander Wang (1): ASoC: SOF: ops: refine parameters order in function snd_sof_dsp_update8 sound/soc/sof/ops.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)

2 1

[PATCH] ASoC: da7219: Improve the IRQ process to increase the stability
by David Rau 16 Feb '23

16 Feb '23

Remove the sleep control in IRQ thread and create an individual task to handel it for Jack plug in event. This commit improves the control of ground switches in the AAD IRQ. Signed-off-by: David Rau <David.Rau.opensource(a)dm.renesas.com> --- sound/soc/codecs/da7219-aad.c | 41 +++++++++++++++++++++++++++++------ sound/soc/codecs/da7219-aad.h | 4 ++++ 2 files changed, 38 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index c55b033d89da..4a4f09f924bc 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -339,11 +339,39 @@ static void da7219_aad_hptest_work(struct work_struct *work) SND_JACK_HEADSET | SND_JACK_LINEOUT); } +static void da7219_aad_jack_det_work(struct work_struct *work) +{ + struct da7219_aad_priv *da7219_aad = + container_of(work, struct da7219_aad_priv, jack_det_work); + struct snd_soc_component *component = da7219_aad->component; + u8 srm_st; + + mutex_lock(&da7219_aad->jack_det_mutex); + + srm_st = snd_soc_component_read(component, DA7219_PLL_SRM_STS) & DA7219_PLL_SRM_STS_MCLK; + msleep(da7219_aad->gnd_switch_delay * ((srm_st == 0x0) ? 2 : 1) - 4); + /* Enable ground switch */ + snd_soc_component_update_bits(component, 0xFB, 0x01, 0x01); + + mutex_unlock(&da7219_aad->jack_det_mutex); +} + /* * IRQ */ +static irqreturn_t da7219_aad_pre_irq_thread(int irq, void *data) +{ + + struct da7219_aad_priv *da7219_aad = data; + + if (!da7219_aad->jack_inserted) + schedule_work(&da7219_aad->jack_det_work); + + return IRQ_WAKE_THREAD; +} + static irqreturn_t da7219_aad_irq_thread(int irq, void *data) { struct da7219_aad_priv *da7219_aad = data; @@ -351,14 +379,9 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data) struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); u8 events[DA7219_AAD_IRQ_REG_MAX]; - u8 statusa, srm_st; + u8 statusa; int i, report = 0, mask = 0; - srm_st = snd_soc_component_read(component, DA7219_PLL_SRM_STS) & DA7219_PLL_SRM_STS_MCLK; - msleep(da7219_aad->gnd_switch_delay * ((srm_st == 0x0) ? 2 : 1) - 4); - /* Enable ground switch */ - snd_soc_component_update_bits(component, 0xFB, 0x01, 0x01); - /* Read current IRQ events */ regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, events, DA7219_AAD_IRQ_REG_MAX); @@ -377,6 +400,9 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data) events[DA7219_AAD_IRQ_REG_A], events[DA7219_AAD_IRQ_REG_B], statusa); + if (!da7219_aad->jack_inserted) + cancel_work_sync(&da7219_aad->jack_det_work); + if (statusa & DA7219_JACK_INSERTION_STS_MASK) { /* Jack Insertion */ if (events[DA7219_AAD_IRQ_REG_A] & @@ -940,8 +966,9 @@ int da7219_aad_init(struct snd_soc_component *component) INIT_WORK(&da7219_aad->btn_det_work, da7219_aad_btn_det_work); INIT_WORK(&da7219_aad->hptest_work, da7219_aad_hptest_work); + INIT_WORK(&da7219_aad->jack_det_work, da7219_aad_jack_det_work); - ret = request_threaded_irq(da7219_aad->irq, NULL, + ret = request_threaded_irq(da7219_aad->irq, da7219_aad_pre_irq_thread, da7219_aad_irq_thread, IRQF_TRIGGER_LOW | IRQF_ONESHOT, "da7219-aad", da7219_aad); diff --git a/sound/soc/codecs/da7219-aad.h b/sound/soc/codecs/da7219-aad.h index 21fdf53095cc..be87ee47edde 100644 --- a/sound/soc/codecs/da7219-aad.h +++ b/sound/soc/codecs/da7219-aad.h @@ -11,6 +11,7 @@ #define __DA7219_AAD_H #include <linux/timer.h> +#include <linux/mutex.h> #include <sound/soc.h> #include <sound/jack.h> #include <sound/da7219-aad.h> @@ -196,6 +197,9 @@ struct da7219_aad_priv { struct work_struct btn_det_work; struct work_struct hptest_work; + struct work_struct jack_det_work; + + struct mutex jack_det_mutex; struct snd_soc_jack *jack; bool micbias_resume_enable; -- 2.17.1

4 5

[PATCHv2] hda/hdmi: Register with vga_switcheroo on Dual GPU Macbooks
by Orlando Chamberlain 16 Feb '23

16 Feb '23

Commit 586bc4aab878 ("ALSA: hda/hdmi - fix vgaswitcheroo detection for AMD") caused only AMD gpu's with PX to have their audio component register with vga_switcheroo. This meant that Apple Macbooks with apple-gmux as the gpu switcher no longer had the audio client registering, so when the gpu is powered off by vga_switcheroo snd_hda_intel is unaware that it should have suspended the device: amdgpu: switched off snd_hda_intel 0000:03:00.1: Unable to change power state from D3hot to D0, device inaccessible snd_hda_intel 0000:03:00.1: CORB reset timeout#2, CORBRP = 65535 To resolve this, we use apple_gmux_detect() and register a vga_switcheroo audio client when apple-gmux is detected. Fixes: 586bc4aab878 ("ALSA: hda/hdmi - fix vgaswitcheroo detection for AMD") Link: https://lore.kernel.org/all/20230210044826.9834-9-orlandoch.dev@gmail.com/ Signed-off-by: Orlando Chamberlain <orlandoch.dev(a)gmail.com> --- v1->v2: Use apple_gmux_detect(), split this patch out of the patch series sound/pci/hda/hda_intel.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 87002670c0c9..cfd2ddfde112 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -50,6 +50,7 @@ #include <sound/intel-dsp-config.h> #include <linux/vgaarb.h> #include <linux/vga_switcheroo.h> +#include <linux/apple-gmux.h> #include <linux/firmware.h> #include <sound/hda_codec.h> #include "hda_controller.h" @@ -1463,7 +1464,7 @@ static struct pci_dev *get_bound_vga(struct pci_dev *pci) * vgaswitcheroo. */ if (((p->class >> 16) == PCI_BASE_CLASS_DISPLAY) && - atpx_present()) + (atpx_present() || apple_gmux_detect(NULL, NULL))) return p; pci_dev_put(p); } -- 2.39.1

2 1

[PATCH v4 00/10] Add the PowerQUICC audio support using the QMC
by Herve Codina 16 Feb '23

16 Feb '23

Hi, This series adds support for audio using the QMC controller available in some Freescale PowerQUICC SoCs. This series contains three parts in order to show the different blocks hierarchy and their usage in this support. The first one is related to TSA (Time Slot Assigner). The TSA handles the data present at the pin level (TDM with up to 64 time slots) and dispatchs them to one or more serial controller (SCC). The second is related to QMC (QUICC Multichannel Controller). The QMC handles the data at the serial controller (SCC) level and splits again the data to creates some virtual channels. The last one is related to the audio component (QMC audio). It is the glue between the QMC controller and the ASoC component. It handles one or more QMC virtual channels and creates one DAI per QMC virtual channels handled. Compared to the previous iteration https://lore.kernel.org/linux-kernel/20230113103759.327698-1-herve.codina@b… this v4 series mainly: - updates code comment format (feedback received on v2 series and forgot to update in v3), - fixes bindings, - Replaces the fsl,tsa phandle and the fsl,tsa-cell-id property by a fsl,tsa-serial phandle and update the TSA related API, - Adds some missing lock in tsa_serial_connect() and tsa_serial_disconnect(). Best regards, Herve Codina Changes v3 -> v4 - patches 2, 6 and 9 Update code comment format. - patch 1 Fix some description formats. Add 'additionalProperties: false' in subnode. Move fsl,mode to fsl,diagnostic-mode. Change clocks and clock-names properties. Add '#serial-cells' property related to the newly introduced fsl,tsa-serial phandle. - patch 2 Move fsl,mode to fsl,diagnostic-mode. Replace the fsl,tsa phandle and the fsl,tsa-cell-id property by a fsl,tsa-serial phandle and update the related API. Add missing locks. - patch 5 Fix some description format. Replace the fsl,tsa phandle and the fsl,tsa-cell-id property by a fsl,tsa-serial phandle. Rename fsl,mode to fsl,operational-mode and update its description. - patch 6 Replace the fsl,tsa phandle and the fsl,tsa-cell-id property by a fsl,tsa-serial phandle and use the TSA updated API. Rename fsl,mode to fsl,operational-mode. - patch 8 Add 'Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>' Changes v2 -> v3 - All bindings Rename fsl-tsa.h to fsl,tsa.h Add missing vendor prefix Various fixes (quotes, node names, upper/lower case) - patches 1 and 2 (TSA binding specific) Remove 'reserved' values in the routing tables Remove fsl,grant-mode Add a better description for 'fsl,common-rxtx-pins' Fix clocks/clocks-name handling against fsl,common-rxtx-pins Add information related to the delays unit Removed FSL_CPM_TSA_NBCELL Fix license in binding header file fsl,tsa.h - patches 5 and 6 (QMC binding specific) Remove fsl,cpm-command property Add interrupt property constraint - patches 8 and 9 (QMC audio binding specific) Remove 'items' in compatible property definition Add missing 'dai-common.yaml' reference Fix the qmc_chan phandle definition - patch 2 and 6 Use io{read,write}be{32,16} Change commit subjects and logs - patch 4 Add 'Acked-by: Christophe Leroy <christophe.leroy(a)csgroup.eu>' Changes v1 -> v2: - patch 2 and 6 Fix kernel test robot errors - other patches No changes Herve Codina (10): dt-bindings: soc: fsl: cpm_qe: Add TSA controller soc: fsl: cpm1: Add support for TSA MAINTAINERS: add the Freescale TSA controller entry powerpc/8xx: Use a larger CPM1 command check mask dt-bindings: soc: fsl: cpm_qe: Add QMC controller soc: fsl: cmp1: Add support for QMC MAINTAINERS: add the Freescale QMC controller entry dt-bindings: sound: Add support for QMC audio ASoC: fsl: Add support for QMC audio MAINTAINERS: add the Freescale QMC audio entry .../bindings/soc/fsl/cpm_qe/fsl,qmc.yaml | 167 ++ .../bindings/soc/fsl/cpm_qe/fsl,tsa.yaml | 261 +++ .../bindings/sound/fsl,qmc-audio.yaml | 117 ++ MAINTAINERS | 25 + arch/powerpc/platforms/8xx/cpm1.c | 2 +- drivers/soc/fsl/qe/Kconfig | 23 + drivers/soc/fsl/qe/Makefile | 2 + drivers/soc/fsl/qe/qmc.c | 1533 +++++++++++++++++ drivers/soc/fsl/qe/tsa.c | 864 ++++++++++ drivers/soc/fsl/qe/tsa.h | 42 + include/dt-bindings/soc/fsl,tsa.h | 13 + include/soc/fsl/qe/qmc.h | 71 + sound/soc/fsl/Kconfig | 9 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_qmc_audio.c | 735 ++++++++ 15 files changed, 3865 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/soc/fsl/cpm_qe/fsl,qmc.yaml create mode 100644 Documentation/devicetree/bindings/soc/fsl/cpm_qe/fsl,tsa.yaml create mode 100644 Documentation/devicetree/bindings/sound/fsl,qmc-audio.yaml create mode 100644 drivers/soc/fsl/qe/qmc.c create mode 100644 drivers/soc/fsl/qe/tsa.c create mode 100644 drivers/soc/fsl/qe/tsa.h create mode 100644 include/dt-bindings/soc/fsl,tsa.h create mode 100644 include/soc/fsl/qe/qmc.h create mode 100644 sound/soc/fsl/fsl_qmc_audio.c -- 2.39.0

6 22

[linux-next:master] BUILD REGRESSION 9d9019bcea1aac7eed64a1a4966282b6b7b141c8
by kernel test robot 16 Feb '23

16 Feb '23

tree/branch: https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git master branch HEAD: 9d9019bcea1aac7eed64a1a4966282b6b7b141c8 Add linux-next specific files for 20230215 Error/Warning reports: https://lore.kernel.org/oe-kbuild-all/202301300743.bp7Dpazv-lkp@intel.com https://lore.kernel.org/oe-kbuild-all/202302061911.C7xvHX9v-lkp@intel.com https://lore.kernel.org/oe-kbuild-all/202302062224.ByzeTXh1-lkp@intel.com https://lore.kernel.org/oe-kbuild-all/202302092211.54EYDhYH-lkp@intel.com https://lore.kernel.org/oe-kbuild-all/202302111601.jtY4lKrA-lkp@intel.com https://lore.kernel.org/oe-kbuild-all/202302112104.g75cGHZd-lkp@intel.com https://lore.kernel.org/oe-kbuild-all/202302142145.iN5WZnpF-lkp@intel.com https://lore.kernel.org/oe-kbuild-all/202302151041.0SWs1RHK-lkp@intel.com https://lore.kernel.org/oe-kbuild-all/202302151109.xlBpZKUS-lkp@intel.com https://lore.kernel.org/oe-kbuild-all/202302151751.DTeft5Fn-lkp@intel.com Error/Warning: (recently discovered and may have been fixed) Documentation/admin-guide/pm/amd-pstate.rst:343: WARNING: duplicate label admin-guide/pm/amd-pstate:user space interface in ``sysfs``, other instance in Documentation/admin-guide/pm/amd-pstate.rst Documentation/riscv/uabi.rst:24: WARNING: Enumerated list ends without a blank line; unexpected unindent. Documentation/sphinx/templates/kernel-toc.html: 1:36 Invalid token: #} ERROR: modpost: "devm_platform_ioremap_resource" [drivers/dma/fsl-edma.ko] undefined! ERROR: modpost: "devm_platform_ioremap_resource" [drivers/dma/idma64.ko] undefined! ERROR: modpost: "walk_hmem_resources" [drivers/dax/hmem/dax_hmem.ko] undefined! arch/mips/kernel/vpe.c:643:41: error: 'struct module' has no member named 'mod_mem' drivers/cxl/core/region.c:2628:6: warning: variable 'rc' is used uninitialized whenever 'if' condition is false [-Wsometimes-uninitialized] drivers/gpu/drm/amd/amdgpu/../display/dc/dcn30/dcn30_optc.c:294:6: warning: no previous prototype for 'optc3_wait_drr_doublebuffer_pending_clear' [-Wmissing-prototypes] drivers/gpu/drm/amd/amdgpu/../display/dc/dcn31/dcn31_hubbub.c:1011:6: warning: no previous prototype for 'hubbub31_init' [-Wmissing-prototypes] drivers/gpu/drm/amd/amdgpu/../display/dc/dcn32/dcn32_hubbub.c:948:6: warning: no previous prototype for 'hubbub32_init' [-Wmissing-prototypes] drivers/gpu/drm/amd/amdgpu/../display/dc/dcn32/dcn32_hubp.c:158:6: warning: no previous prototype for 'hubp32_init' [-Wmissing-prototypes] drivers/gpu/drm/amd/amdgpu/../display/dc/dcn32/dcn32_resource_helpers.c:62:18: warning: variable 'cursor_bpp' set but not used [-Wunused-but-set-variable] drivers/gpu/drm/amd/amdgpu/../display/dc/link/link_detection.c:1199: warning: expecting prototype for dc_link_detect_connection_type(). Prototype was for link_detect_connection_type() instead drivers/gpu/drm/amd/amdgpu/../display/dc/link/protocols/link_dp_capability.c:1292:32: warning: variable 'result_write_min_hblank' set but not used [-Wunused-but-set-variable] drivers/gpu/drm/amd/amdgpu/../display/dc/link/protocols/link_dp_training.c:1586:38: warning: variable 'result' set but not used [-Wunused-but-set-variable] idma64.c:(.text+0x7c): undefined reference to `devm_platform_ioremap_resource' Unverified Error/Warning (likely false positive, please contact us if interested): drivers/clk/ingenic/jz4760-cgu.c:80 jz4760_cgu_calc_m_n_od() error: uninitialized symbol 'od'. drivers/hid/hid-sony.c:1726:41: warning: Uninitialized variables: entry.lock, entry.list_node, entry.hdev, entry.touchpad, entry.sensor_dev, entry.quirks, entry.state_worker, entry.send_output_report, entry.battery, entry.battery_desc, entry.device_id, entry.output_report_dmabuf, entry.state_worker_initialized, entry.defer_initialization, entry.battery_capacity, entry.battery_status, entry.led_count, entry.ghl_urb, entry.ghl_poke_timer [uninitvar] drivers/usb/gadget/composite.c:2082:33: sparse: sparse: restricted __le16 degrades to integer drivers/usb/host/xhci-plat.c:371 xhci_generic_plat_probe() error: we previously assumed 'sysdev' could be null (see line 361) sound/soc/soc-ac97.c:339 snd_soc_ac97_parse_pinctl() warn: passing zero to 'PTR_ERR' Error/Warning ids grouped by kconfigs: gcc_recent_errors |-- alpha-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-..-display-dc-link-link_detection.c:warning:expecting-prototype-for-dc_link_detect_connection_type().-Prototype-was-for-link_detect_connection_type()-instead | |-- drivers-gpu-drm-amd-amdgpu-..-display-dc-link-protocols-link_dp_capability.c:warning:variable-result_write_min_hblank-set-but-not-used | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-link-protocols-link_dp_training.c:warning:variable-result-set-but-not-used |-- alpha-randconfig-r033-20230212 | |-- drivers-gpu-drm-amd-amdgpu-..-display-dc-link-link_detection.c:warning:expecting-prototype-for-dc_link_detect_connection_type().-Prototype-was-for-link_detect_connection_type()-instead | |-- drivers-gpu-drm-amd-amdgpu-..-display-dc-link-protocols-link_dp_capability.c:warning:variable-result_write_min_hblank-set-but-not-used | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-link-protocols-link_dp_training.c:warning:variable-result-set-but-not-used |-- arc-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-..-display-dc-link-link_detection.c:warning:expecting-prototype-for-dc_link_detect_connection_type().-Prototype-was-for-link_detect_connection_type()-instead | |-- drivers-gpu-drm-amd-amdgpu-..-display-dc-link-protocols-link_dp_capability.c:warning:variable-result_write_min_hblank-set-but-not-used | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-link-protocols-link_dp_training.c:warning:variable-result-set-but-not-used |-- arm-allmodconfig | |-- drivers-gpu-drm-amd-amdgpu-..-display-dc-link-link_detection.c:warning:expecting-prototype-for-dc_link_detect_connection_type().-Prototype-was-for-link_detect_connection_type()-instead | |-- drivers-gpu-drm-amd-amdgpu-..-display-dc-link-protocols-link_dp_capability.c:warning:variable-result_write_min_hblank-set-but-not-used | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-link-protocols-link_dp_training.c:warning:variable-result-set-but-not-used |-- arm-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-..-display-dc-link-link_detection.c:warning:expecting-prototype-for-dc_link_detect_connection_type().-Prototype-was-for-link_detect_connection_type()-instead | |-- drivers-gpu-drm-amd-amdgpu-..-display-dc-link-protocols-link_dp_capability.c:warning:variable-result_write_min_hblank-set-but-not-used | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-link-protocols-link_dp_training.c:warning:variable-result-set-but-not-used |-- arm64-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dcn30-dcn30_optc.c:warning:no-previous-prototype-for-optc3_wait_drr_doublebuffer_pending_clear | |-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dcn31-dcn31_hubbub.c:warning:no-previous-prototype-for-hubbub31_init | |-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dcn32-dcn32_hubbub.c:warning:no-previous-prototype-for-hubbub32_init | |-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dcn32-dcn32_hubp.c:warning:no-previous-prototype-for-hubp32_init | |-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dcn32-dcn32_resource_helpers.c:warning:variable-cursor_bpp-set-but-not-used | |-- drivers-gpu-drm-amd-amdgpu-..-display-dc-link-link_detection.c:warning:expecting-prototype-for-dc_link_detect_connection_type().-Prototype-was-for-link_detect_connection_type()-instead | |-- drivers-gpu-drm-amd-amdgpu-..-display-dc-link-protocols-link_dp_capability.c:warning:variable-result_write_min_hblank-set-but-not-used | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-link-protocols-link_dp_training.c:warning:variable-result-set-but-not-used |-- i386-allyesconfig | |-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dcn30-dcn30_optc.c:warning:no-previous-prototype-for-optc3_wait_drr_doublebuffer_pending_clear | |-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dcn31-dcn31_hubbub.c:warning:no-previous-prototype-for-hubbub31_init | |-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dcn32-dcn32_hubbub.c:warning:no-previous-prototype-for-hubbub32_init | |-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dcn32-dcn32_hubp.c:warning:no-previous-prototype-for-hubp32_init | |-- drivers-gpu-drm-amd-amdgpu-..-display-dc-dcn32-dcn32_resource_helpers.c:warning:variable-cursor_bpp-set-but-not-used | |-- drivers-gpu-drm-amd-amdgpu-..-display-dc-link-link_detection.c:warning:expecting-prototype-for-dc_link_detect_connection_type().-Prototype-was-for-link_detect_connection_type()-instead | |-- drivers-gpu-drm-amd-amdgpu-..-display-dc-link-protocols-link_dp_capability.c:warning:variable-result_write_min_hblank-set-but-not-used | `-- drivers-gpu-drm-amd-amdgpu-..-display-dc-link-protocols-link_dp_training.c:warning:variable-result-set-but-not-used |-- i386-randconfig-c001 | `-- net-sched-act_pedit.c:WARNING-opportunity-for-kmemdup |-- i386-randconfig-s001 | |-- drivers-gpu-drm-i915-gem-i915_gem_ttm.c:sparse:sparse:incorrect-type-in-assignment-(different-base-types)-expected-restricted-vm_fault_t-assigned-usertype-ret-got-int | `-- drivers-usb-gadget-composite.c:sparse:sparse:restricted-__le16-degrades-to-integer |-- i386-randconfig-s002 | `-- drivers-gpu-drm-i915-gem-i915_gem_ttm.c:sparse:sparse:incorrect-type-in-assignment-(different-base-types)-expected-restricted-vm_fault_t-assigned-usertype-ret-got-int |-- i386-randconfig-s003 | `-- drivers-usb-gadget-composite.c:sparse:sparse:restricted-__le16-degrades-to-integer |-- ia64-allmodconfig | |-- drivers-gpu-drm-amd-amdgpu-..-display-dc-link-link_detection.c:warning:expecting-prototype-for-dc_link_detect_connection_type().-Prototype-was-for-link_detect_connection_type()-instead | |-- drivers-gpu-drm-amd-amdgpu-..-display-dc-link-protocols-link_dp_capability.c:warning:variable-result_write_min_hblank-set-but-not-used clang_recent_errors |-- arm-randconfig-r005-20230212 | `-- drivers-cxl-core-region.c:warning:variable-rc-is-used-uninitialized-whenever-if-condition-is-false |-- s390-randconfig-r001-20230213 | `-- idma64.c:(.text):undefined-reference-to-devm_platform_ioremap_resource |-- x86_64-randconfig-a002-20230213 | `-- drivers-cxl-core-region.c:warning:variable-rc-is-used-uninitialized-whenever-if-condition-is-false |-- x86_64-randconfig-a004-20230213 | `-- drivers-cxl-core-region.c:warning:variable-rc-is-used-uninitialized-whenever-if-condition-is-false `-- x86_64-randconfig-a005-20230213 `-- drivers-cxl-core-region.c:warning:variable-rc-is-used-uninitialized-whenever-if-condition-is-false elapsed time: 1294m configs tested: 108 configs skipped: 7 gcc tested configs: alpha allyesconfig alpha defconfig arc allyesconfig arc defconfig arc nsim_700_defconfig arc randconfig-r043-20230212 arc randconfig-r043-20230213 arm allmodconfig arm allyesconfig arm defconfig arm randconfig-r046-20230212 arm u8500_defconfig arm64 allyesconfig arm64 defconfig csky defconfig i386 allyesconfig i386 debian-10.3 i386 defconfig i386 randconfig-a011-20230213 i386 randconfig-a012-20230213 i386 randconfig-a013-20230213 i386 randconfig-a014-20230213 i386 randconfig-a015-20230213 i386 randconfig-a016-20230213 i386 randconfig-c001 ia64 allmodconfig ia64 defconfig loongarch allmodconfig loongarch allnoconfig loongarch defconfig m68k allmodconfig m68k allyesconfig m68k defconfig m68k mvme147_defconfig m68k sun3x_defconfig mips allmodconfig mips allyesconfig mips ar7_defconfig mips decstation_r4k_defconfig mips gcw0_defconfig mips xway_defconfig nios2 allyesconfig nios2 defconfig openrisc defconfig parisc defconfig parisc64 defconfig powerpc allmodconfig powerpc allnoconfig powerpc allyesconfig powerpc mgcoge_defconfig powerpc stx_gp3_defconfig powerpc tqm8548_defconfig riscv allmodconfig riscv allnoconfig riscv allyesconfig riscv defconfig riscv randconfig-r042-20230213 riscv rv32_defconfig s390 allmodconfig s390 allyesconfig s390 defconfig s390 randconfig-r044-20230213 sh allmodconfig sh dreamcast_defconfig sh rsk7264_defconfig sh se7712_defconfig sparc defconfig um i386_defconfig um x86_64_defconfig x86_64 allnoconfig x86_64 allyesconfig x86_64 defconfig x86_64 kexec x86_64 randconfig-a011-20230213 x86_64 randconfig-a012-20230213 x86_64 randconfig-a013-20230213 x86_64 randconfig-a014-20230213 x86_64 randconfig-a015-20230213 x86_64 randconfig-a016-20230213 x86_64 rhel-8.3 xtensa allyesconfig xtensa smp_lx200_defconfig clang tested configs: arm multi_v5_defconfig arm randconfig-r046-20230213 hexagon randconfig-r041-20230212 hexagon randconfig-r041-20230213 hexagon randconfig-r045-20230212 hexagon randconfig-r045-20230213 i386 randconfig-a001-20230213 i386 randconfig-a002-20230213 i386 randconfig-a003-20230213 i386 randconfig-a004-20230213 i386 randconfig-a005-20230213 i386 randconfig-a006-20230213 mips malta_kvm_defconfig mips randconfig-c004-20230212 powerpc mpc8313_rdb_defconfig powerpc ppc44x_defconfig powerpc randconfig-c003-20230212 riscv randconfig-r042-20230212 s390 randconfig-r044-20230212 x86_64 randconfig-a001-20230213 x86_64 randconfig-a002-20230213 x86_64 randconfig-a003-20230213 x86_64 randconfig-a004-20230213 x86_64 randconfig-a005-20230213 x86_64 randconfig-a006-20230213 x86_64 randconfig-k001 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests

1 0

[PATCH v2] ASoC: soc-dapm.h: fixup warning struct snd_pcm_substream not declared
by Lucas Tanure 16 Feb '23

16 Feb '23

Add struct snd_pcm_substream forward declaration Fixes: 078a85f2806f ("ASoC: dapm: Only power up active channels from a DAI") Signed-off-by: Lucas Tanure <lucas.tanure(a)collabora.com> --- Changes since v1: - Use forward declaration instead of header sound/pcm.h --- include/sound/soc-dapm.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 77495e5988c1..64915ebd641e 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -16,6 +16,7 @@ #include <sound/asoc.h> struct device; +struct snd_pcm_substream; struct snd_soc_pcm_runtime; struct soc_enum; -- 2.39.1

4 3

[PATCH v1 1/1] ASoC: soc-ac97: Return correct error codes
by Andy Shevchenko 16 Feb '23

16 Feb '23

With the switching to dev_err_probe(), during the conversion of GPIO calls, the return code is passed is a paratemer to it. At the same time a copy'n'paste mistake was made, so the wrong variable has been taken for the error reporting. Fix this. Fixes: 3ee0d39c50dc ("ASoC: soc-ac97: Convert to agnostic GPIO API") Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <error27(a)gmail.com> Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/soc-ac97.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index 40051f513c6a..4e4fe29ade50 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -336,12 +336,12 @@ static int snd_soc_ac97_parse_pinctl(struct device *dev, cfg->sdata_gpio = devm_gpiod_get_index(dev, "ac97", 1, GPIOD_ASIS); if (IS_ERR(cfg->sdata_gpio)) - return dev_err_probe(dev, PTR_ERR(cfg->sync_gpio), "Can't find ac97-sdata gpio\n"); + return dev_err_probe(dev, PTR_ERR(cfg->sdata_gpio), "Can't find ac97-sdata gpio\n"); gpiod_set_consumer_name(cfg->sdata_gpio, "AC97 link sdata"); cfg->reset_gpio = devm_gpiod_get_index(dev, "ac97", 2, GPIOD_ASIS); if (IS_ERR(cfg->reset_gpio)) - return dev_err_probe(dev, PTR_ERR(cfg->sync_gpio), "Can't find ac97-reset gpio\n"); + return dev_err_probe(dev, PTR_ERR(cfg->reset_gpio), "Can't find ac97-reset gpio\n"); gpiod_set_consumer_name(cfg->reset_gpio, "AC97 link reset"); return 0; -- 2.39.1

2 1

[PATCH v2 1/2] ASoC: cs35l45: Remove separate tables module
by Charles Keepax 16 Feb '23

16 Feb '23

There is no reason to have a separate module for the tables file it just holds regmap callbacks and register patches used by the main part of the driver. Remove the separate module and merge it into the main driver module. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- Changes since v1: - Remove module macros as the bottom of cs35l45-tables.c sound/soc/codecs/Kconfig | 5 ----- sound/soc/codecs/Makefile | 4 +--- sound/soc/codecs/cs35l45-tables.c | 4 ---- 3 files changed, 1 insertion(+), 12 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4621674e68bf1..bd72c426a93d1 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -688,9 +688,6 @@ config SND_SOC_CS35L41_I2C select SND_SOC_CS35L41 select REGMAP_I2C -config SND_SOC_CS35L45_TABLES - tristate - config SND_SOC_CS35L45 tristate @@ -699,7 +696,6 @@ config SND_SOC_CS35L45_SPI depends on SPI_MASTER select REGMAP select REGMAP_SPI - select SND_SOC_CS35L45_TABLES select SND_SOC_CS35L45 help Enable support for Cirrus Logic CS35L45 smart speaker amplifier @@ -710,7 +706,6 @@ config SND_SOC_CS35L45_I2C depends on I2C select REGMAP select REGMAP_I2C - select SND_SOC_CS35L45_TABLES select SND_SOC_CS35L45 help Enable support for Cirrus Logic CS35L45 smart speaker amplifier diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a0a61554548eb..f1ca18f7946ca 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -63,8 +63,7 @@ snd-soc-cs35l41-lib-objs := cs35l41-lib.o snd-soc-cs35l41-objs := cs35l41.o snd-soc-cs35l41-spi-objs := cs35l41-spi.o snd-soc-cs35l41-i2c-objs := cs35l41-i2c.o -snd-soc-cs35l45-tables-objs := cs35l45-tables.o -snd-soc-cs35l45-objs := cs35l45.o +snd-soc-cs35l45-objs := cs35l45.o cs35l45-tables.o snd-soc-cs35l45-spi-objs := cs35l45-spi.o snd-soc-cs35l45-i2c-objs := cs35l45-i2c.o snd-soc-cs42l42-objs := cs42l42.o @@ -431,7 +430,6 @@ obj-$(CONFIG_SND_SOC_CS35L41) += snd-soc-cs35l41.o obj-$(CONFIG_SND_SOC_CS35L41_LIB) += snd-soc-cs35l41-lib.o obj-$(CONFIG_SND_SOC_CS35L41_SPI) += snd-soc-cs35l41-spi.o obj-$(CONFIG_SND_SOC_CS35L41_I2C) += snd-soc-cs35l41-i2c.o -obj-$(CONFIG_SND_SOC_CS35L45_TABLES) += snd-soc-cs35l45-tables.o obj-$(CONFIG_SND_SOC_CS35L45) += snd-soc-cs35l45.o obj-$(CONFIG_SND_SOC_CS35L45_SPI) += snd-soc-cs35l45-spi.o obj-$(CONFIG_SND_SOC_CS35L45_I2C) += snd-soc-cs35l45-i2c.o diff --git a/sound/soc/codecs/cs35l45-tables.c b/sound/soc/codecs/cs35l45-tables.c index 5a2c2e684ef9f..5a2f47ca57716 100644 --- a/sound/soc/codecs/cs35l45-tables.c +++ b/sound/soc/codecs/cs35l45-tables.c @@ -196,7 +196,3 @@ unsigned int cs35l45_get_clk_freq_id(unsigned int freq) return -EINVAL; } EXPORT_SYMBOL_NS_GPL(cs35l45_get_clk_freq_id, SND_SOC_CS35L45_TABLES); - -MODULE_DESCRIPTION("ASoC CS35L45 driver tables"); -MODULE_AUTHOR("James Schulman, Cirrus Logic Inc, <james.schulman(a)cirrus.com>"); -MODULE_LICENSE("Dual BSD/GPL"); -- 2.30.2

2 2

[Patch] [cs42l42] Remove unneeded version.h include pointed out by 'make versioncheck'
by Jesper Juhl 16 Feb '23

16 Feb '23

From 80dd7af06ef675c183bc7faedc474e5cdda295cb Mon Sep 17 00:00:00 2001 From: Jesper Juhl <jesperjuhl76(a)gmail.com> Date: Mon, 13 Feb 2023 02:54:45 +0100 Subject: [PATCH 08/12] [cs42l42] Remove unneeded version.h include pointed out by 'make versioncheck' Signed-off-by: Jesper Juhl <jesperjuhl76(a)gmail.com> --- sound/soc/codecs/cs42l42.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 2fefbcf7bd13..ba8c81b6ac87 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -11,7 +11,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/types.h> #include <linux/init.h> #include <linux/delay.h> -- 2.39.2

1 0

[PATCH AUTOSEL 5.15 08/12] ASoC: rt715-sdca: fix clock stop prepare timeout issue
by Sasha Levin 15 Feb '23

15 Feb '23

From: Jack Yu <jack.yu(a)realtek.com> [ Upstream commit 2036890282d56bcbf7f915ba9e04bf77967ab231 ] Modify clock_stop_timeout value for rt715-sdca according to the requirement of internal clock trimming. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> Link: https://lore.kernel.org/r/574b6586267a458cac78c5ac4d5b10bd@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt715-sdca-sdw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt715-sdca-sdw.c b/sound/soc/codecs/rt715-sdca-sdw.c index 0f4354eafef25..85abf8073c278 100644 --- a/sound/soc/codecs/rt715-sdca-sdw.c +++ b/sound/soc/codecs/rt715-sdca-sdw.c @@ -167,7 +167,7 @@ static int rt715_sdca_read_prop(struct sdw_slave *slave) } /* set the timeout values */ - prop->clk_stop_timeout = 20; + prop->clk_stop_timeout = 200; return 0; } -- 2.39.0

1 0

[PATCH AUTOSEL 6.1 20/24] ASoC: codecs: es8326: Fix DTS properties reading
by Sasha Levin 15 Feb '23

15 Feb '23

From: Alexey Firago <a.firago(a)yadro.com> [ Upstream commit fe1e7e8ce2c47bd8fd9885eab63fca0a522e94c9 ] Seems like properties parsing and reading was copy-pasted, so "everest,interrupt-src" and "everest,interrupt-clk" are saved into the es8326->jack_pol variable. This might lead to wrong settings being saved into the reg 57 (ES8326_HP_DET). Fix this by using proper variables while reading properties. Signed-off-by: Alexey Firago <a.firago(a)yadro.com> Reviewed-by: Yang Yingliang <yangyingliang(a)huawei.com Link: https://lore.kernel.org/r/20230204195106.46539-1-a.firago@yadro.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8326.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index 87c1cc16592bb..555125efd9ad3 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -729,14 +729,16 @@ static int es8326_probe(struct snd_soc_component *component) } dev_dbg(component->dev, "jack-pol %x", es8326->jack_pol); - ret = device_property_read_u8(component->dev, "everest,interrupt-src", &es8326->jack_pol); + ret = device_property_read_u8(component->dev, "everest,interrupt-src", + &es8326->interrupt_src); if (ret != 0) { dev_dbg(component->dev, "interrupt-src return %d", ret); es8326->interrupt_src = ES8326_HP_DET_SRC_PIN9; } dev_dbg(component->dev, "interrupt-src %x", es8326->interrupt_src); - ret = device_property_read_u8(component->dev, "everest,interrupt-clk", &es8326->jack_pol); + ret = device_property_read_u8(component->dev, "everest,interrupt-clk", + &es8326->interrupt_clk); if (ret != 0) { dev_dbg(component->dev, "interrupt-clk return %d", ret); es8326->interrupt_clk = 0x45; -- 2.39.0

1 0

[PATCH AUTOSEL 6.1 16/24] ASoC: SOF: amd: Fix for handling spurious interrupts from DSP
by Sasha Levin 15 Feb '23

15 Feb '23

From: V sujith kumar Reddy <Vsujithkumar.Reddy(a)amd.com> [ Upstream commit 2e7c6652f9b86c01cbd4e988057a746a3a461969 ] As interrupts are Level-triggered,unless and until we deassert the register the interrupts are generated which causes spurious interrupts unhandled. Now we deasserted the interrupt at top half which solved the below "nobody cared" warning. warning reported in dmesg: irq 80: nobody cared (try booting with the "irqpoll" option) CPU: 5 PID: 2735 Comm: irq/80-AudioDSP Not tainted 5.15.86-15817-g4c19f3e06d49 #1 1bd3fd932cf58caacc95b0504d6ea1e3eab22289 Hardware name: Google Skyrim/Skyrim, BIOS Google_Skyrim.15303.0.0 01/03/2023 Call Trace: <IRQ> dump_stack_lvl+0x69/0x97 __report_bad_irq+0x3a/0xae note_interrupt+0x1a9/0x1e3 handle_irq_event_percpu+0x4b/0x6e handle_irq_event+0x36/0x5b handle_fasteoi_irq+0xae/0x171 __common_interrupt+0x48/0xc4 </IRQ> handlers: acp_irq_handler [snd_sof_amd_acp] threaded [<000000007e089f34>] acp_irq_thread [snd_sof_amd_acp] Disabling IRQ #80 Signed-off-by: V sujith kumar Reddy <Vsujithkumar.Reddy(a)amd.com> Link: https://lore.kernel.org/r/20230203123254.1898794-1-Vsujithkumar.Reddy@amd.c… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/amd/acp.c | 36 +++++++++++++++--------------------- 1 file changed, 15 insertions(+), 21 deletions(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 36966643e36ab..8afd67ba1e5a3 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -316,7 +316,6 @@ static irqreturn_t acp_irq_thread(int irq, void *context) { struct snd_sof_dev *sdev = context; const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); - unsigned int base = desc->dsp_intr_base; unsigned int val, count = ACP_HW_SEM_RETRY_COUNT; val = snd_sof_dsp_read(sdev, ACP_DSP_BAR, desc->ext_intr_stat); @@ -326,28 +325,20 @@ static irqreturn_t acp_irq_thread(int irq, void *context) return IRQ_HANDLED; } - val = snd_sof_dsp_read(sdev, ACP_DSP_BAR, base + DSP_SW_INTR_STAT_OFFSET); - if (val & ACP_DSP_TO_HOST_IRQ) { - while (snd_sof_dsp_read(sdev, ACP_DSP_BAR, desc->hw_semaphore_offset)) { - /* Wait until acquired HW Semaphore lock or timeout */ - count--; - if (!count) { - dev_err(sdev->dev, "%s: Failed to acquire HW lock\n", __func__); - return IRQ_NONE; - } + while (snd_sof_dsp_read(sdev, ACP_DSP_BAR, desc->hw_semaphore_offset)) { + /* Wait until acquired HW Semaphore lock or timeout */ + count--; + if (!count) { + dev_err(sdev->dev, "%s: Failed to acquire HW lock\n", __func__); + return IRQ_NONE; } - - sof_ops(sdev)->irq_thread(irq, sdev); - val |= ACP_DSP_TO_HOST_IRQ; - snd_sof_dsp_write(sdev, ACP_DSP_BAR, base + DSP_SW_INTR_STAT_OFFSET, val); - - /* Unlock or Release HW Semaphore */ - snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->hw_semaphore_offset, 0x0); - - return IRQ_HANDLED; } - return IRQ_NONE; + sof_ops(sdev)->irq_thread(irq, sdev); + /* Unlock or Release HW Semaphore */ + snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->hw_semaphore_offset, 0x0); + + return IRQ_HANDLED; }; static irqreturn_t acp_irq_handler(int irq, void *dev_id) @@ -358,8 +349,11 @@ static irqreturn_t acp_irq_handler(int irq, void *dev_id) unsigned int val; val = snd_sof_dsp_read(sdev, ACP_DSP_BAR, base + DSP_SW_INTR_STAT_OFFSET); - if (val) + if (val) { + val |= ACP_DSP_TO_HOST_IRQ; + snd_sof_dsp_write(sdev, ACP_DSP_BAR, base + DSP_SW_INTR_STAT_OFFSET, val); return IRQ_WAKE_THREAD; + } return IRQ_NONE; } -- 2.39.0

1 0

[PATCH AUTOSEL 6.1 13/24] ASoC: rt715-sdca: fix clock stop prepare timeout issue
by Sasha Levin 15 Feb '23

15 Feb '23

From: Jack Yu <jack.yu(a)realtek.com> [ Upstream commit 2036890282d56bcbf7f915ba9e04bf77967ab231 ] Modify clock_stop_timeout value for rt715-sdca according to the requirement of internal clock trimming. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> Link: https://lore.kernel.org/r/574b6586267a458cac78c5ac4d5b10bd@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt715-sdca-sdw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt715-sdca-sdw.c b/sound/soc/codecs/rt715-sdca-sdw.c index 3f981a9e7fb67..c54ecf3e69879 100644 --- a/sound/soc/codecs/rt715-sdca-sdw.c +++ b/sound/soc/codecs/rt715-sdca-sdw.c @@ -167,7 +167,7 @@ static int rt715_sdca_read_prop(struct sdw_slave *slave) } /* set the timeout values */ - prop->clk_stop_timeout = 20; + prop->clk_stop_timeout = 200; return 0; } -- 2.39.0

1 0

[PATCH 1/2] dt-bindings: sound: ep93xx: Add I2S and AC'97 descriptions
by Alexander Sverdlin 15 Feb '23

15 Feb '23

Add device tree bindings for Cirrus Logic EP93xx internal SoCs' I2S and AC'97 controllers. Signed-off-by: Alexander Sverdlin <alexander.sverdlin(a)gmail.com> --- .../bindings/sound/cirrus,ep9301-ac97.yaml | 47 ++++++++++++++ .../bindings/sound/cirrus,ep9301-i2s.yaml | 63 +++++++++++++++++++ MAINTAINERS | 1 + 3 files changed, 111 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,ep9301-ac97.yaml create mode 100644 Documentation/devicetree/bindings/sound/cirrus,ep9301-i2s.yaml diff --git a/Documentation/devicetree/bindings/sound/cirrus,ep9301-ac97.yaml b/Documentation/devicetree/bindings/sound/cirrus,ep9301-ac97.yaml new file mode 100644 index 000000000000..bfa5d392f525 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,ep9301-ac97.yaml @@ -0,0 +1,47 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,ep9301-ac97.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus EP93xx AC'97 Controller + +description: | + The AC’97 Controller includes a 5-pin serial interface to an external audio + codec. + +maintainers: + - Alexander Sverdlin <alexander.sverdlin(a)gmail.com> + +properties: + compatible: + const: cirrus,ep9301-ac97 + + '#sound-dai-cells': + const: 0 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + +required: + - compatible + - '#sound-dai-cells' + - reg + - interrupts + +additionalProperties: false + +examples: + - | + ac97: ac97@80880000 { + compatible = "cirrus,ep9301-ac97"; + #sound-dai-cells = <0>; + reg = <0x80880000 0xac>; + interrupt-parent = <&vic0>; + interrupts = <6>; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/cirrus,ep9301-i2s.yaml b/Documentation/devicetree/bindings/sound/cirrus,ep9301-i2s.yaml new file mode 100644 index 000000000000..b47d28dc8389 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,ep9301-i2s.yaml @@ -0,0 +1,63 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,ep9301-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus EP93xx I2S Controller + +description: | + The I2S controller is used to stream serial audio data between the external + I2S CODECs’, ADCs/DACs, and the ARM Core. The controller supports I2S, Left- + and Right-Justified DSP formats. + +maintainers: + - Alexander Sverdlin <alexander.sverdlin(a)gmail.com> + +properties: + compatible: + const: cirrus,ep9301-i2s + + '#sound-dai-cells': + const: 0 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + minItems: 3 + + clock-names: + items: + - const: mclk + - const: sclk + - const: lrclk + +required: + - compatible + - '#sound-dai-cells' + - reg + - interrupts + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + i2s: i2s@80820000 { + compatible = "cirrus,ep9301-i2s"; + #sound-dai-cells = <0>; + reg = <0x80820000 0x100>; + interrupt-parent = <&vic1>; + interrupts = <28>; + clocks = <&syscon 29 + &syscon 30 + &syscon 31>; + clock-names = "mclk", "sclk", "lrclk"; + }; + +... diff --git a/MAINTAINERS b/MAINTAINERS index 254aab631fd4..765902756c37 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -2092,6 +2092,7 @@ M: Alexander Sverdlin <alexander.sverdlin(a)gmail.com> L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) S: Maintained F: Documentation/devicetree/bindings/iio/adc/cirrus,ep9301-adc.yaml +F: Documentation/devicetree/bindings/sound/cirrus,ep9301-* F: arch/arm/mach-ep93xx/ F: arch/arm/mach-ep93xx/include/mach/ F: drivers/iio/adc/ep93xx_adc.c -- 2.39.1

4 8

[PATCH] ASoC: soc-dapm.h: fixup warning struct snd_pcm_substream not declared
by Lucas Tanure 15 Feb '23

15 Feb '23

Add sound/pcm.h header to include struct snd_pcm_substream used in snd_soc_dapm_update_dai function. Fixes: 078a85f2806f ("ASoC: dapm: Only power up active channels from a DAI") Signed-off-by: Lucas Tanure <lucas.tanure(a)collabora.com> --- include/sound/soc-dapm.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 77495e5988c1..8fbb50a11d80 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -14,6 +14,7 @@ #include <sound/control.h> #include <sound/soc-topology.h> #include <sound/asoc.h> +#include <sound/pcm.h> struct device; struct snd_soc_pcm_runtime; -- 2.39.1

4 3

[PATCH] ASoC: mediatek: mt8188: correct etdm control return value
by Trevor Wu 15 Feb '23

15 Feb '23

In mt8188_etdm_clk_src_sel_put() function, val retrieved by FIELD_PREP is shifted to the corresponding bit filed, so it can compare with the register value directly. Originally, the redundant bit shift of the register value results in the wrong comparison result, so we remove bit shift operation in the patch. Fixes: 2babb4777489 ("ASoC: mediatek: mt8188: support etdm in platform driver") Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> --- sound/soc/mediatek/mt8188/mt8188-dai-etdm.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/mediatek/mt8188/mt8188-dai-etdm.c b/sound/soc/mediatek/mt8188/mt8188-dai-etdm.c index 071841903c62..1c53d4cb19bb 100644 --- a/sound/soc/mediatek/mt8188/mt8188-dai-etdm.c +++ b/sound/soc/mediatek/mt8188/mt8188-dai-etdm.c @@ -679,7 +679,6 @@ static int mt8188_etdm_clk_src_sel_put(struct snd_kcontrol *kcontrol, unsigned int old_val; unsigned int mask; unsigned int reg; - unsigned int shift; if (source >= e->items) return -EINVAL; @@ -687,27 +686,22 @@ static int mt8188_etdm_clk_src_sel_put(struct snd_kcontrol *kcontrol, if (!strcmp(kcontrol->id.name, "ETDM_OUT1_Clock_Source")) { reg = ETDM_OUT1_CON4; mask = ETDM_OUT_CON4_CLOCK_MASK; - shift = ETDM_OUT_CON4_CLOCK_SHIFT; val = FIELD_PREP(ETDM_OUT_CON4_CLOCK_MASK, source); } else if (!strcmp(kcontrol->id.name, "ETDM_OUT2_Clock_Source")) { reg = ETDM_OUT2_CON4; mask = ETDM_OUT_CON4_CLOCK_MASK; - shift = ETDM_OUT_CON4_CLOCK_SHIFT; val = FIELD_PREP(ETDM_OUT_CON4_CLOCK_MASK, source); } else if (!strcmp(kcontrol->id.name, "ETDM_OUT3_Clock_Source")) { reg = ETDM_OUT3_CON4; mask = ETDM_OUT_CON4_CLOCK_MASK; - shift = ETDM_OUT_CON4_CLOCK_SHIFT; val = FIELD_PREP(ETDM_OUT_CON4_CLOCK_MASK, source); } else if (!strcmp(kcontrol->id.name, "ETDM_IN1_Clock_Source")) { reg = ETDM_IN1_CON2; mask = ETDM_IN_CON2_CLOCK_MASK; - shift = ETDM_IN_CON2_CLOCK_SHIFT; val = FIELD_PREP(ETDM_IN_CON2_CLOCK_MASK, source); } else if (!strcmp(kcontrol->id.name, "ETDM_IN2_Clock_Source")) { reg = ETDM_IN2_CON2; mask = ETDM_IN_CON2_CLOCK_MASK; - shift = ETDM_IN_CON2_CLOCK_SHIFT; val = FIELD_PREP(ETDM_IN_CON2_CLOCK_MASK, source); } else { return -EINVAL; @@ -715,8 +709,6 @@ static int mt8188_etdm_clk_src_sel_put(struct snd_kcontrol *kcontrol, regmap_read(afe->regmap, reg, &old_val); old_val &= mask; - old_val >>= shift; - if (old_val == val) return 0; -- 2.18.0

1 0

[PATCH] ASoC: cs35l45: Remove separate tables module
by Charles Keepax 15 Feb '23

15 Feb '23

There is no reason to have a separate module for the tables file it just holds regmap callbacks and register patches used by the main part of the driver. Remove the separate module and merge it into the main driver module. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/Kconfig | 5 ----- sound/soc/codecs/Makefile | 4 +--- 2 files changed, 1 insertion(+), 8 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4621674e68bf1..bd72c426a93d1 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -688,9 +688,6 @@ config SND_SOC_CS35L41_I2C select SND_SOC_CS35L41 select REGMAP_I2C -config SND_SOC_CS35L45_TABLES - tristate - config SND_SOC_CS35L45 tristate @@ -699,7 +696,6 @@ config SND_SOC_CS35L45_SPI depends on SPI_MASTER select REGMAP select REGMAP_SPI - select SND_SOC_CS35L45_TABLES select SND_SOC_CS35L45 help Enable support for Cirrus Logic CS35L45 smart speaker amplifier @@ -710,7 +706,6 @@ config SND_SOC_CS35L45_I2C depends on I2C select REGMAP select REGMAP_I2C - select SND_SOC_CS35L45_TABLES select SND_SOC_CS35L45 help Enable support for Cirrus Logic CS35L45 smart speaker amplifier diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a0a61554548eb..f1ca18f7946ca 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -63,8 +63,7 @@ snd-soc-cs35l41-lib-objs := cs35l41-lib.o snd-soc-cs35l41-objs := cs35l41.o snd-soc-cs35l41-spi-objs := cs35l41-spi.o snd-soc-cs35l41-i2c-objs := cs35l41-i2c.o -snd-soc-cs35l45-tables-objs := cs35l45-tables.o -snd-soc-cs35l45-objs := cs35l45.o +snd-soc-cs35l45-objs := cs35l45.o cs35l45-tables.o snd-soc-cs35l45-spi-objs := cs35l45-spi.o snd-soc-cs35l45-i2c-objs := cs35l45-i2c.o snd-soc-cs42l42-objs := cs42l42.o @@ -431,7 +430,6 @@ obj-$(CONFIG_SND_SOC_CS35L41) += snd-soc-cs35l41.o obj-$(CONFIG_SND_SOC_CS35L41_LIB) += snd-soc-cs35l41-lib.o obj-$(CONFIG_SND_SOC_CS35L41_SPI) += snd-soc-cs35l41-spi.o obj-$(CONFIG_SND_SOC_CS35L41_I2C) += snd-soc-cs35l41-i2c.o -obj-$(CONFIG_SND_SOC_CS35L45_TABLES) += snd-soc-cs35l45-tables.o obj-$(CONFIG_SND_SOC_CS35L45) += snd-soc-cs35l45.o obj-$(CONFIG_SND_SOC_CS35L45_SPI) += snd-soc-cs35l45-spi.o obj-$(CONFIG_SND_SOC_CS35L45_I2C) += snd-soc-cs35l45-i2c.o -- 2.30.2

1 1

[PATCH v1 1/1] ASoC: soc-ac97: Convert to agnostic GPIO API
by Andy Shevchenko 15 Feb '23

15 Feb '23

The of_gpio.h is going to be removed. In preparation of that convert the driver to the agnostic API. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/soc-ac97.c | 68 ++++++++++++++------------------------------ 1 file changed, 22 insertions(+), 46 deletions(-) diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index 32c5be61e2ec..40051f513c6a 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -14,10 +14,9 @@ #include <linux/ctype.h> #include <linux/delay.h> #include <linux/export.h> -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/gpio/driver.h> #include <linux/init.h> -#include <linux/of_gpio.h> #include <linux/of.h> #include <linux/pinctrl/consumer.h> #include <linux/slab.h> @@ -29,9 +28,9 @@ struct snd_ac97_reset_cfg { struct pinctrl_state *pstate_reset; struct pinctrl_state *pstate_warm_reset; struct pinctrl_state *pstate_run; - int gpio_sdata; - int gpio_sync; - int gpio_reset; + struct gpio_desc *reset_gpio; + struct gpio_desc *sdata_gpio; + struct gpio_desc *sync_gpio; }; static struct snd_ac97_bus soc_ac97_bus = { @@ -268,11 +267,11 @@ static void snd_soc_ac97_warm_reset(struct snd_ac97 *ac97) pinctrl_select_state(pctl, snd_ac97_rst_cfg.pstate_warm_reset); - gpio_direction_output(snd_ac97_rst_cfg.gpio_sync, 1); + gpiod_direction_output_raw(snd_ac97_rst_cfg.sync_gpio, 1); udelay(10); - gpio_direction_output(snd_ac97_rst_cfg.gpio_sync, 0); + gpiod_direction_output_raw(snd_ac97_rst_cfg.sync_gpio, 0); pinctrl_select_state(pctl, snd_ac97_rst_cfg.pstate_run); msleep(2); @@ -284,13 +283,13 @@ static void snd_soc_ac97_reset(struct snd_ac97 *ac97) pinctrl_select_state(pctl, snd_ac97_rst_cfg.pstate_reset); - gpio_direction_output(snd_ac97_rst_cfg.gpio_sync, 0); - gpio_direction_output(snd_ac97_rst_cfg.gpio_sdata, 0); - gpio_direction_output(snd_ac97_rst_cfg.gpio_reset, 0); + gpiod_direction_output_raw(snd_ac97_rst_cfg.sync_gpio, 0); + gpiod_direction_output_raw(snd_ac97_rst_cfg.sdata_gpio, 0); + gpiod_direction_output_raw(snd_ac97_rst_cfg.reset_gpio, 0); udelay(10); - gpio_direction_output(snd_ac97_rst_cfg.gpio_reset, 1); + gpiod_direction_output_raw(snd_ac97_rst_cfg.reset_gpio, 1); pinctrl_select_state(pctl, snd_ac97_rst_cfg.pstate_run); msleep(2); @@ -301,8 +300,6 @@ static int snd_soc_ac97_parse_pinctl(struct device *dev, { struct pinctrl *p; struct pinctrl_state *state; - int gpio; - int ret; p = devm_pinctrl_get(dev); if (IS_ERR(p)) { @@ -332,41 +329,20 @@ static int snd_soc_ac97_parse_pinctl(struct device *dev, } cfg->pstate_run = state; - gpio = of_get_named_gpio(dev->of_node, "ac97-gpios", 0); - if (gpio < 0) { - dev_err(dev, "Can't find ac97-sync gpio\n"); - return gpio; - } - ret = devm_gpio_request(dev, gpio, "AC97 link sync"); - if (ret) { - dev_err(dev, "Failed requesting ac97-sync gpio\n"); - return ret; - } - cfg->gpio_sync = gpio; + cfg->sync_gpio = devm_gpiod_get_index(dev, "ac97", 0, GPIOD_ASIS); + if (IS_ERR(cfg->sync_gpio)) + return dev_err_probe(dev, PTR_ERR(cfg->sync_gpio), "Can't find ac97-sync gpio\n"); + gpiod_set_consumer_name(cfg->sync_gpio, "AC97 link sync"); - gpio = of_get_named_gpio(dev->of_node, "ac97-gpios", 1); - if (gpio < 0) { - dev_err(dev, "Can't find ac97-sdata gpio %d\n", gpio); - return gpio; - } - ret = devm_gpio_request(dev, gpio, "AC97 link sdata"); - if (ret) { - dev_err(dev, "Failed requesting ac97-sdata gpio\n"); - return ret; - } - cfg->gpio_sdata = gpio; + cfg->sdata_gpio = devm_gpiod_get_index(dev, "ac97", 1, GPIOD_ASIS); + if (IS_ERR(cfg->sdata_gpio)) + return dev_err_probe(dev, PTR_ERR(cfg->sync_gpio), "Can't find ac97-sdata gpio\n"); + gpiod_set_consumer_name(cfg->sdata_gpio, "AC97 link sdata"); - gpio = of_get_named_gpio(dev->of_node, "ac97-gpios", 2); - if (gpio < 0) { - dev_err(dev, "Can't find ac97-reset gpio\n"); - return gpio; - } - ret = devm_gpio_request(dev, gpio, "AC97 link reset"); - if (ret) { - dev_err(dev, "Failed requesting ac97-reset gpio\n"); - return ret; - } - cfg->gpio_reset = gpio; + cfg->reset_gpio = devm_gpiod_get_index(dev, "ac97", 2, GPIOD_ASIS); + if (IS_ERR(cfg->reset_gpio)) + return dev_err_probe(dev, PTR_ERR(cfg->sync_gpio), "Can't find ac97-reset gpio\n"); + gpiod_set_consumer_name(cfg->reset_gpio, "AC97 link reset"); return 0; } -- 2.39.1

2 1

[PATCH V2 2/8] soundwire: amd: Add support for AMD Manager driver
by Vijendar Mukunda 15 Feb '23

15 Feb '23

AMD ACP(v6.x) IP block has two soundwire manager devices. Add support for - Manager driver probe & remove sequence - Helper functions to enable/disable interrupts, Initialize sdw manager, enable sdw pads - Manager driver sdw_master_ops & port_ops callbacks Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 761 ++++++++++++++++++++++++++++++ drivers/soundwire/amd_manager.h | 252 ++++++++++ include/linux/soundwire/sdw_amd.h | 67 +++ 3 files changed, 1080 insertions(+) create mode 100644 drivers/soundwire/amd_manager.c create mode 100644 drivers/soundwire/amd_manager.h create mode 100644 include/linux/soundwire/sdw_amd.h diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c new file mode 100644 index 000000000000..cd1e5a3d5995 --- /dev/null +++ b/drivers/soundwire/amd_manager.c @@ -0,0 +1,761 @@ +// SPDX-License-Identifier: GPL-2.0+ +/* + * SoundWire AMD Manager driver + * + * Copyright 2023 Advanced Micro Devices, Inc. + */ + +#include <linux/completion.h> +#include <linux/device.h> +#include <linux/io.h> +#include <linux/jiffies.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/wait.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "bus.h" +#include "amd_manager.h" + +#define DRV_NAME "amd_sdw_manager" + +#define to_amd_sdw(b) container_of(b, struct amd_sdw_manager, bus) + +static void amd_enable_sdw_pads(struct amd_sdw_manager *amd_manager) +{ + u32 sw_pad_pulldown_val; + u32 val = 0; + + mutex_lock(amd_manager->sdw_lock); + val = acp_reg_readl(amd_manager->acp_mmio + ACP_SW_PAD_KEEPER_EN); + val |= amd_manager->reg_mask->sw_pad_enable_mask; + acp_reg_writel(val, amd_manager->acp_mmio + ACP_SW_PAD_KEEPER_EN); + usleep_range(1000, 1500); + + sw_pad_pulldown_val = acp_reg_readl(amd_manager->acp_mmio + ACP_PAD_PULLDOWN_CTRL); + sw_pad_pulldown_val &= amd_manager->reg_mask->sw_pad_pulldown_mask; + acp_reg_writel(sw_pad_pulldown_val, amd_manager->acp_mmio + ACP_PAD_PULLDOWN_CTRL); + mutex_unlock(amd_manager->sdw_lock); +} + +static int amd_init_sdw_manager(struct amd_sdw_manager *amd_manager) +{ + u32 val = 0; + u32 timeout = 0; + u32 retry_count = 0; + + acp_reg_writel(AMD_SDW_ENABLE, amd_manager->mmio + ACP_SW_EN); + do { + val = acp_reg_readl(amd_manager->mmio + ACP_SW_EN_STATUS); + if (val) + break; + usleep_range(10, 50); + } while (retry_count++ < AMD_SDW_STAT_MAX_RETRY_COUNT); + + if (retry_count > AMD_SDW_STAT_MAX_RETRY_COUNT) + return -ETIMEDOUT; + + /* Soundwire manager bus reset */ + acp_reg_writel(AMD_SDW_BUS_RESET_REQ, amd_manager->mmio + ACP_SW_BUS_RESET_CTRL); + val = acp_reg_readl(amd_manager->mmio + ACP_SW_BUS_RESET_CTRL); + while (!(val & AMD_SDW_BUS_RESET_DONE)) { + val = acp_reg_readl(amd_manager->mmio + ACP_SW_BUS_RESET_CTRL); + if (timeout > AMD_DELAY_LOOP_ITERATION) + break; + usleep_range(1, 5); + timeout++; + } + if (timeout == AMD_DELAY_LOOP_ITERATION) + return -ETIMEDOUT; + timeout = 0; + acp_reg_writel(AMD_SDW_BUS_RESET_CLEAR_REQ, amd_manager->mmio + ACP_SW_BUS_RESET_CTRL); + val = acp_reg_readl(amd_manager->mmio + ACP_SW_BUS_RESET_CTRL); + while (val) { + val = acp_reg_readl(amd_manager->mmio + ACP_SW_BUS_RESET_CTRL); + if (timeout > AMD_DELAY_LOOP_ITERATION) + break; + usleep_range(1, 5); + timeout++; + } + if (timeout == AMD_DELAY_LOOP_ITERATION) { + dev_err(amd_manager->dev, "Failed to reset Soundwire manager instance%d\n", + amd_manager->instance); + return -ETIMEDOUT; + } + retry_count = 0; + acp_reg_writel(AMD_SDW_DISABLE, amd_manager->mmio + ACP_SW_EN); + do { + val = acp_reg_readl(amd_manager->mmio + ACP_SW_EN_STATUS); + if (!val) + break; + usleep_range(10, 50); + } while (retry_count++ < AMD_SDW_STAT_MAX_RETRY_COUNT); + + if (retry_count > AMD_SDW_STAT_MAX_RETRY_COUNT) + return -ETIMEDOUT; + return 0; +} + +static int amd_enable_sdw_manager(struct amd_sdw_manager *amd_manager) +{ + u32 val = 0; + u32 retry_count = 0; + + acp_reg_writel(AMD_SDW_ENABLE, amd_manager->mmio + ACP_SW_EN); + do { + val = acp_reg_readl(amd_manager->mmio + ACP_SW_EN_STATUS); + if (val) + break; + usleep_range(10, 50); + } while (retry_count++ < AMD_SDW_STAT_MAX_RETRY_COUNT); + + if (retry_count > AMD_SDW_STAT_MAX_RETRY_COUNT) + return -ETIMEDOUT; + return 0; +} + +static int amd_disable_sdw_manager(struct amd_sdw_manager *amd_manager) +{ + u32 val = 0; + u32 retry_count = 0; + + acp_reg_writel(AMD_SDW_DISABLE, amd_manager->mmio + ACP_SW_EN); + /* + * After invoking manager disable sequence, check whether + * manager has executed clock stop sequence. In this case, + * manager should ignore checking enable status register. + */ + val = acp_reg_readl(amd_manager->mmio + ACP_SW_CLK_RESUME_CTRL); + if (val) + return 0; + + do { + val = acp_reg_readl(amd_manager->mmio + ACP_SW_EN_STATUS); + if (!val) + break; + usleep_range(10, 50); + } while (retry_count++ < AMD_SDW_STAT_MAX_RETRY_COUNT); + + if (retry_count > AMD_SDW_STAT_MAX_RETRY_COUNT) + return -ETIMEDOUT; + return 0; +} + +static void amd_enable_sdw_interrupts(struct amd_sdw_manager *amd_manager) +{ + struct sdw_manager_reg_mask *reg_mask = amd_manager->reg_mask; + u32 val; + + mutex_lock(amd_manager->sdw_lock); + val = acp_reg_readl(amd_manager->acp_mmio + ACP_EXTERNAL_INTR_CNTL(amd_manager->instance)); + val |= reg_mask->acp_sdw_intr_mask; + acp_reg_writel(val, amd_manager->acp_mmio + ACP_EXTERNAL_INTR_CNTL(amd_manager->instance)); + val = acp_reg_readl(amd_manager->acp_mmio + ACP_EXTERNAL_INTR_CNTL(amd_manager->instance)); + mutex_unlock(amd_manager->sdw_lock); + dev_dbg(amd_manager->dev, "%s: acp_ext_intr_ctrl[0x%x]:0x%x\n", __func__, + ACP_EXTERNAL_INTR_CNTL(amd_manager->instance), val); + val = acp_reg_readl(amd_manager->acp_mmio + ACP_EXTERNAL_INTR_STAT(amd_manager->instance)); + if (val) + acp_reg_writel(val, amd_manager->acp_mmio + + ACP_EXTERNAL_INTR_STAT(amd_manager->instance)); + + acp_reg_writel(AMD_SDW_IRQ_MASK_0TO7, amd_manager->mmio + + ACP_SW_STATE_CHANGE_STATUS_MASK_0TO7); + acp_reg_writel(AMD_SDW_IRQ_MASK_8TO11, amd_manager->mmio + + ACP_SW_STATE_CHANGE_STATUS_MASK_8TO11); + acp_reg_writel(AMD_SDW_IRQ_ERROR_MASK, amd_manager->mmio + ACP_SW_ERROR_INTR_MASK); +} + +static void amd_disable_sdw_interrupts(struct amd_sdw_manager *amd_manager) +{ + struct sdw_manager_reg_mask *reg_mask = amd_manager->reg_mask; + u32 val; + + mutex_lock(amd_manager->sdw_lock); + val = acp_reg_readl(amd_manager->acp_mmio + ACP_EXTERNAL_INTR_CNTL(amd_manager->instance)); + val &= ~reg_mask->acp_sdw_intr_mask; + acp_reg_writel(val, amd_manager->acp_mmio + ACP_EXTERNAL_INTR_CNTL(amd_manager->instance)); + mutex_unlock(amd_manager->sdw_lock); + + acp_reg_writel(0x00, amd_manager->mmio + ACP_SW_STATE_CHANGE_STATUS_MASK_0TO7); + acp_reg_writel(0x00, amd_manager->mmio + ACP_SW_STATE_CHANGE_STATUS_MASK_8TO11); + acp_reg_writel(0x00, amd_manager->mmio + ACP_SW_ERROR_INTR_MASK); +} + +static void amd_sdw_set_frameshape(struct amd_sdw_manager *amd_manager) +{ + u32 frame_size; + + frame_size = (amd_manager->rows_index << 3) | amd_manager->cols_index; + acp_reg_writel(frame_size, amd_manager->mmio + ACP_SW_FRAMESIZE); +} + +static void amd_sdw_ctl_word_prep(u32 *low_word, u32 *high_word, u32 cmd_type, + struct sdw_msg *msg, int cmd_offset) +{ + u32 low_data = 0, high_data = 0; + u16 addr; + u8 addr_high, addr_low; + u8 data = 0; + + addr = msg->addr + cmd_offset; + addr_high = (addr & 0xFF00) >> 8; + addr_low = addr & 0xFF; + + if (cmd_type == AMD_SDW_CMD_WRITE) + data = msg->buf[cmd_offset]; + + high_data = FIELD_PREP(AMD_SDW_MCP_CMD_DEV_ADDR, msg->dev_num); + high_data |= FIELD_PREP(AMD_SDW_MCP_CMD_COMMAND, cmd_type); + high_data |= FIELD_PREP(AMD_SDW_MCP_CMD_REG_ADDR_HIGH, addr_high); + low_data |= FIELD_PREP(AMD_SDW_MCP_CMD_REG_ADDR_LOW, addr_low); + low_data |= FIELD_PREP(AMD_SDW_MCP_CMD_REG_DATA, data); + + *high_word = high_data; + *low_word = low_data; +} + +static u64 amd_sdw_send_cmd_get_resp(struct amd_sdw_manager *amd_manager, u32 lword, u32 uword) +{ + u64 resp = 0; + u32 resp_lower, resp_high; + u32 sts = 0; + u32 timeout = 0; + + sts = acp_reg_readl(amd_manager->mmio + ACP_SW_IMM_CMD_STS); + while (sts & AMD_SDW_IMM_CMD_BUSY) { + sts = acp_reg_readl(amd_manager->mmio + ACP_SW_IMM_CMD_STS); + if (timeout > AMD_SDW_RETRY_COUNT) { + dev_err(amd_manager->dev, "SDW%x previous cmd status clear failed\n", + amd_manager->instance); + return -ETIMEDOUT; + } + timeout++; + } + + timeout = 0; + if (sts & AMD_SDW_IMM_RES_VALID) { + dev_err(amd_manager->dev, "SDW%x manager is in bad state\n", amd_manager->instance); + acp_reg_writel(0x00, amd_manager->mmio + ACP_SW_IMM_CMD_STS); + } + acp_reg_writel(uword, amd_manager->mmio + ACP_SW_IMM_CMD_UPPER_WORD); + acp_reg_writel(lword, amd_manager->mmio + ACP_SW_IMM_CMD_LOWER_QWORD); + + sts = acp_reg_readl(amd_manager->mmio + ACP_SW_IMM_CMD_STS); + while (!(sts & AMD_SDW_IMM_RES_VALID)) { + sts = acp_reg_readl(amd_manager->mmio + ACP_SW_IMM_CMD_STS); + if (timeout > AMD_SDW_RETRY_COUNT) { + dev_err(amd_manager->dev, "SDW%x cmd response timeout occurred\n", + amd_manager->instance); + return -ETIMEDOUT; + } + timeout++; + } + resp_high = acp_reg_readl(amd_manager->mmio + ACP_SW_IMM_RESP_UPPER_WORD); + resp_lower = acp_reg_readl(amd_manager->mmio + ACP_SW_IMM_RESP_LOWER_QWORD); + timeout = 0; + acp_reg_writel(AMD_SDW_IMM_RES_VALID, amd_manager->mmio + ACP_SW_IMM_CMD_STS); + while ((sts & AMD_SDW_IMM_RES_VALID)) { + sts = acp_reg_readl(amd_manager->mmio + ACP_SW_IMM_CMD_STS); + if (timeout > AMD_SDW_RETRY_COUNT) { + dev_err(amd_manager->dev, "SDW%x cmd status retry failed\n", + amd_manager->instance); + return -ETIMEDOUT; + } + timeout++; + } + resp = resp_high; + resp = (resp << 32) | resp_lower; + return resp; +} + +static enum sdw_command_response +amd_program_scp_addr(struct amd_sdw_manager *amd_manager, struct sdw_msg *msg) +{ + struct sdw_msg scp_msg = {0}; + u64 response_buf[2] = {0}; + u32 uword = 0, lword = 0; + int nack = 0, no_ack = 0; + int index, timeout = 0; + + scp_msg.dev_num = msg->dev_num; + scp_msg.addr = SDW_SCP_ADDRPAGE1; + scp_msg.buf = &msg->addr_page1; + amd_sdw_ctl_word_prep(&lword, &uword, AMD_SDW_CMD_WRITE, &scp_msg, 0); + response_buf[0] = amd_sdw_send_cmd_get_resp(amd_manager, lword, uword); + scp_msg.addr = SDW_SCP_ADDRPAGE2; + scp_msg.buf = &msg->addr_page2; + amd_sdw_ctl_word_prep(&lword, &uword, AMD_SDW_CMD_WRITE, &scp_msg, 0); + response_buf[1] = amd_sdw_send_cmd_get_resp(amd_manager, lword, uword); + + /* check response the writes */ + for (index = 0; index < 2; index++) { + if (response_buf[index] == -ETIMEDOUT) { + dev_err(amd_manager->dev, "Program SCP cmd timeout\n"); + timeout = 1; + } else if (!(response_buf[index] & AMD_SDW_MCP_RESP_ACK)) { + no_ack = 1; + if (response_buf[index] & AMD_SDW_MCP_RESP_NACK) { + nack = 1; + dev_err(amd_manager->dev, "Program SCP NACK received\n"); + } + } + } + + if (timeout) { + dev_err_ratelimited(amd_manager->dev, + "SCP_addrpage command timeout for Slave %d\n", msg->dev_num); + return SDW_CMD_TIMEOUT; + } + + if (nack) { + dev_err_ratelimited(amd_manager->dev, + "SCP_addrpage NACKed for Slave %d\n", msg->dev_num); + return SDW_CMD_FAIL; + } + + if (no_ack) { + dev_dbg_ratelimited(amd_manager->dev, + "SCP_addrpage ignored for Slave %d\n", msg->dev_num); + return SDW_CMD_IGNORED; + } + return SDW_CMD_OK; +} + +static int amd_prep_msg(struct amd_sdw_manager *amd_manager, struct sdw_msg *msg, int *cmd) +{ + int ret; + + if (msg->page) { + ret = amd_program_scp_addr(amd_manager, msg); + if (ret) { + msg->len = 0; + return ret; + } + } + switch (msg->flags) { + case SDW_MSG_FLAG_READ: + *cmd = AMD_SDW_CMD_READ; + break; + case SDW_MSG_FLAG_WRITE: + *cmd = AMD_SDW_CMD_WRITE; + break; + default: + dev_err(amd_manager->dev, "Invalid msg cmd: %d\n", msg->flags); + return -EINVAL; + } + return 0; +} + +static enum sdw_command_response amd_sdw_fill_msg_resp(struct amd_sdw_manager *amd_manager, + struct sdw_msg *msg, u64 response, + int offset) +{ + int nack = 0, no_ack = 0; + int timeout = 0; + + if (response & AMD_SDW_MCP_RESP_ACK) { + if (msg->flags == SDW_MSG_FLAG_READ) + msg->buf[offset] = FIELD_GET(AMD_SDW_MCP_RESP_RDATA, response); + } else { + no_ack = 1; + if (response == -ETIMEDOUT) { + timeout = 1; + } else if (response & AMD_SDW_MCP_RESP_NACK) { + nack = 1; + dev_err(amd_manager->dev, "Program SCP NACK received\n"); + } + } + + if (timeout) { + dev_err_ratelimited(amd_manager->dev, "command timeout for Slave %d\n", + msg->dev_num); + return SDW_CMD_TIMEOUT; + } + if (nack) { + dev_err_ratelimited(amd_manager->dev, + "command response NACK received for Slave %d\n", msg->dev_num); + return SDW_CMD_FAIL; + } + + if (no_ack) { + dev_err_ratelimited(amd_manager->dev, "command is ignored for Slave %d\n", + msg->dev_num); + return SDW_CMD_IGNORED; + } + return SDW_CMD_OK; +} + +static unsigned int _amd_sdw_xfer_msg(struct amd_sdw_manager *amd_manager, struct sdw_msg *msg, + int cmd, int cmd_offset) +{ + u64 response = 0; + u32 uword = 0, lword = 0; + + amd_sdw_ctl_word_prep(&lword, &uword, cmd, msg, cmd_offset); + response = amd_sdw_send_cmd_get_resp(amd_manager, lword, uword); + return amd_sdw_fill_msg_resp(amd_manager, msg, response, cmd_offset); +} + +static enum sdw_command_response amd_sdw_xfer_msg(struct sdw_bus *bus, struct sdw_msg *msg) +{ + struct amd_sdw_manager *amd_manager = to_amd_sdw(bus); + int ret, i; + int cmd = 0; + + ret = amd_prep_msg(amd_manager, msg, &cmd); + if (ret) + return SDW_CMD_FAIL_OTHER; + for (i = 0; i < msg->len; i++) { + ret = _amd_sdw_xfer_msg(amd_manager, msg, cmd, i); + if (ret) + return ret; + } + return SDW_CMD_OK; +} + +static u32 amd_sdw_read_ping_status(struct sdw_bus *bus) +{ + struct amd_sdw_manager *amd_manager = to_amd_sdw(bus); + u64 response; + u32 slave_stat = 0; + + response = amd_sdw_send_cmd_get_resp(amd_manager, 0, 0); + /* slave status from ping response*/ + slave_stat = FIELD_GET(AMD_SDW_MCP_SLAVE_STAT_0_3, response); + slave_stat |= FIELD_GET(AMD_SDW_MCP_SLAVE_STAT_4_11, response) << 8; + dev_dbg(amd_manager->dev, "%s: slave_stat:0x%x\n", __func__, slave_stat); + return slave_stat; +} + +static int amd_sdw_compute_params(struct sdw_bus *bus) +{ + struct sdw_transport_data t_data = {0}; + struct sdw_master_runtime *m_rt; + struct sdw_port_runtime *p_rt; + struct sdw_bus_params *b_params = &bus->params; + int port_bo, hstart, hstop, sample_int; + unsigned int rate, bps; + + port_bo = 0; + hstart = 1; + hstop = bus->params.col - 1; + t_data.hstop = hstop; + t_data.hstart = hstart; + + list_for_each_entry(m_rt, &bus->m_rt_list, bus_node) { + rate = m_rt->stream->params.rate; + bps = m_rt->stream->params.bps; + sample_int = (bus->params.curr_dr_freq / rate); + list_for_each_entry(p_rt, &m_rt->port_list, port_node) { + port_bo = (p_rt->num * 64) + 1; + dev_dbg(bus->dev, "p_rt->num=%d hstart=%d hstop=%d port_bo=%d\n", + p_rt->num, hstart, hstop, port_bo); + sdw_fill_xport_params(&p_rt->transport_params, p_rt->num, + false, SDW_BLK_GRP_CNT_1, sample_int, + port_bo, port_bo >> 8, hstart, hstop, + SDW_BLK_PKG_PER_PORT, 0x0); + + sdw_fill_port_params(&p_rt->port_params, + p_rt->num, bps, + SDW_PORT_FLOW_MODE_ISOCH, + b_params->m_data_mode); + t_data.hstart = hstart; + t_data.hstop = hstop; + t_data.block_offset = port_bo; + t_data.sub_block_offset = 0; + } + sdw_compute_slave_ports(m_rt, &t_data); + } + return 0; +} + +static int amd_sdw_port_params(struct sdw_bus *bus, struct sdw_port_params *p_params, + unsigned int bank) +{ + struct amd_sdw_manager *amd_manager = to_amd_sdw(bus); + u32 frame_fmt_reg, dpn_frame_fmt; + + dev_dbg(amd_manager->dev, "%s: p_params->num:0x%x\n", __func__, p_params->num); + switch (amd_manager->instance) { + case ACP_SDW0: + frame_fmt_reg = sdw0_manager_dp_reg[p_params->num].frame_fmt_reg; + break; + case ACP_SDW1: + frame_fmt_reg = sdw1_manager_dp_reg[p_params->num].frame_fmt_reg; + break; + default: + return -EINVAL; + } + + dpn_frame_fmt = acp_reg_readl(amd_manager->mmio + frame_fmt_reg); + u32p_replace_bits(&dpn_frame_fmt, p_params->flow_mode, AMD_DPN_FRAME_FMT_PFM); + u32p_replace_bits(&dpn_frame_fmt, p_params->data_mode, AMD_DPN_FRAME_FMT_PDM); + u32p_replace_bits(&dpn_frame_fmt, p_params->bps - 1, AMD_DPN_FRAME_FMT_WORD_LEN); + acp_reg_writel(dpn_frame_fmt, amd_manager->mmio + frame_fmt_reg); + return 0; +} + +static int amd_sdw_transport_params(struct sdw_bus *bus, + struct sdw_transport_params *params, + enum sdw_reg_bank bank) +{ + struct amd_sdw_manager *amd_manager = to_amd_sdw(bus); + u32 dpn_frame_fmt; + u32 dpn_sampleinterval; + u32 dpn_hctrl; + u32 dpn_offsetctrl; + u32 dpn_lanectrl; + u32 frame_fmt_reg, sample_int_reg, hctrl_dp0_reg; + u32 offset_reg, lane_ctrl_ch_en_reg; + + switch (amd_manager->instance) { + case ACP_SDW0: + frame_fmt_reg = sdw0_manager_dp_reg[params->port_num].frame_fmt_reg; + sample_int_reg = sdw0_manager_dp_reg[params->port_num].sample_int_reg; + hctrl_dp0_reg = sdw0_manager_dp_reg[params->port_num].hctrl_dp0_reg; + offset_reg = sdw0_manager_dp_reg[params->port_num].offset_reg; + lane_ctrl_ch_en_reg = sdw0_manager_dp_reg[params->port_num].lane_ctrl_ch_en_reg; + break; + case ACP_SDW1: + frame_fmt_reg = sdw1_manager_dp_reg[params->port_num].frame_fmt_reg; + sample_int_reg = sdw1_manager_dp_reg[params->port_num].sample_int_reg; + hctrl_dp0_reg = sdw1_manager_dp_reg[params->port_num].hctrl_dp0_reg; + offset_reg = sdw1_manager_dp_reg[params->port_num].offset_reg; + lane_ctrl_ch_en_reg = sdw1_manager_dp_reg[params->port_num].lane_ctrl_ch_en_reg; + break; + default: + return -EINVAL; + } + acp_reg_writel(AMD_SDW_SSP_COUNTER_VAL, amd_manager->mmio + ACP_SW_SSP_COUNTER); + + dpn_frame_fmt = acp_reg_readl(amd_manager->mmio + frame_fmt_reg); + u32p_replace_bits(&dpn_frame_fmt, params->blk_pkg_mode, AMD_DPN_FRAME_FMT_BLK_PKG_MODE); + u32p_replace_bits(&dpn_frame_fmt, params->blk_grp_ctrl, AMD_DPN_FRAME_FMT_BLK_GRP_CTRL); + u32p_replace_bits(&dpn_frame_fmt, SDW_STREAM_PCM, AMD_DPN_FRAME_FMT_PCM_OR_PDM); + acp_reg_writel(dpn_frame_fmt, amd_manager->mmio + frame_fmt_reg); + + dpn_sampleinterval = params->sample_interval - 1; + acp_reg_writel(dpn_sampleinterval, amd_manager->mmio + sample_int_reg); + + dpn_hctrl = FIELD_PREP(AMD_DPN_HCTRL_HSTOP, params->hstop); + dpn_hctrl |= FIELD_PREP(AMD_DPN_HCTRL_HSTART, params->hstart); + acp_reg_writel(dpn_hctrl, amd_manager->mmio + hctrl_dp0_reg); + + dpn_offsetctrl = FIELD_PREP(AMD_DPN_OFFSET_CTRL_1, params->offset1); + dpn_offsetctrl |= FIELD_PREP(AMD_DPN_OFFSET_CTRL_2, params->offset2); + acp_reg_writel(dpn_offsetctrl, amd_manager->mmio + offset_reg); + + /* + * lane_ctrl_ch_en_reg will be used to program lane_ctrl and ch_mask + * parameters. + */ + dpn_lanectrl = acp_reg_readl(amd_manager->mmio + lane_ctrl_ch_en_reg); + u32p_replace_bits(&dpn_lanectrl, params->lane_ctrl, AMD_DPN_CH_EN_LCTRL); + acp_reg_writel(dpn_lanectrl, amd_manager->mmio + lane_ctrl_ch_en_reg); + return 0; +} + +static int amd_sdw_port_enable(struct sdw_bus *bus, + struct sdw_enable_ch *enable_ch, + unsigned int bank) +{ + struct amd_sdw_manager *amd_manager = to_amd_sdw(bus); + u32 dpn_ch_enable; + u32 lane_ctrl_ch_en_reg; + + switch (amd_manager->instance) { + case ACP_SDW0: + lane_ctrl_ch_en_reg = sdw0_manager_dp_reg[enable_ch->port_num].lane_ctrl_ch_en_reg; + break; + case ACP_SDW1: + lane_ctrl_ch_en_reg = sdw1_manager_dp_reg[enable_ch->port_num].lane_ctrl_ch_en_reg; + break; + default: + return -EINVAL; + } + + /* + * lane_ctrl_ch_en_reg will be used to program lane_ctrl and ch_mask + * parameters. + */ + dpn_ch_enable = acp_reg_readl(amd_manager->mmio + lane_ctrl_ch_en_reg); + u32p_replace_bits(&dpn_ch_enable, enable_ch->ch_mask, AMD_DPN_CH_EN_CHMASK); + if (enable_ch->enable) + acp_reg_writel(dpn_ch_enable, amd_manager->mmio + lane_ctrl_ch_en_reg); + else + acp_reg_writel(0, amd_manager->mmio + lane_ctrl_ch_en_reg); + return 0; +} + +static int sdw_master_read_amd_prop(struct sdw_bus *bus) +{ + struct amd_sdw_manager *amd_manager = to_amd_sdw(bus); + struct fwnode_handle *link; + struct sdw_master_prop *prop; + u32 quirk_mask = 0; + u32 wake_en_mask = 0; + u32 power_mode_mask = 0; + char name[32]; + + prop = &bus->prop; + /* Find manager handle */ + snprintf(name, sizeof(name), "mipi-sdw-link-%d-subproperties", bus->link_id); + link = device_get_named_child_node(bus->dev, name); + if (!link) { + dev_err(bus->dev, "Manager node %s not found\n", name); + return -EIO; + } + fwnode_property_read_u32(link, "amd-sdw-enable", &quirk_mask); + if (!(quirk_mask & AMD_SDW_QUIRK_MASK_BUS_ENABLE)) + prop->hw_disabled = true; + prop->quirks = SDW_MASTER_QUIRKS_CLEAR_INITIAL_CLASH | + SDW_MASTER_QUIRKS_CLEAR_INITIAL_PARITY; + + fwnode_property_read_u32(link, "amd-sdw-wake-enable", &wake_en_mask); + amd_manager->wake_en_mask = wake_en_mask; + fwnode_property_read_u32(link, "amd-sdw-power-mode", &power_mode_mask); + amd_manager->power_mode_mask = power_mode_mask; + return 0; +} + +static int amd_prop_read(struct sdw_bus *bus) +{ + sdw_master_read_prop(bus); + sdw_master_read_amd_prop(bus); + return 0; +} + +static const struct sdw_master_port_ops amd_sdw_port_ops = { + .dpn_set_port_params = amd_sdw_port_params, + .dpn_set_port_transport_params = amd_sdw_transport_params, + .dpn_port_enable_ch = amd_sdw_port_enable, +}; + +static const struct sdw_master_ops amd_sdw_ops = { + .read_prop = amd_prop_read, + .xfer_msg = amd_sdw_xfer_msg, + .read_ping_status = amd_sdw_read_ping_status, +}; + +static void amd_sdw_probe_work(struct work_struct *work) +{ + struct amd_sdw_manager *amd_manager = container_of(work, struct amd_sdw_manager, + probe_work); + struct sdw_master_prop *prop; + int ret; + + prop = &amd_manager->bus.prop; + if (!prop->hw_disabled) { + amd_enable_sdw_pads(amd_manager); + ret = amd_init_sdw_manager(amd_manager); + if (ret) + return; + amd_enable_sdw_interrupts(amd_manager); + ret = amd_enable_sdw_manager(amd_manager); + if (ret) + return; + amd_sdw_set_frameshape(amd_manager); + } +} + +static int amd_sdw_manager_probe(struct platform_device *pdev) +{ + const struct acp_sdw_pdata *pdata = pdev->dev.platform_data; + struct resource *res; + struct device *dev = &pdev->dev; + struct sdw_master_prop *prop; + struct sdw_bus_params *params; + struct amd_sdw_manager *amd_manager; + int ret; + + if (!pdev->dev.platform_data) { + dev_err(dev, "platform_data not retrieved\n"); + return -ENODEV; + } + amd_manager = devm_kzalloc(dev, sizeof(struct amd_sdw_manager), GFP_KERNEL); + if (!amd_manager) + return -ENOMEM; + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) + return -ENOMEM; + amd_manager->acp_mmio = devm_ioremap(dev, res->start, resource_size(res)); + if (IS_ERR(amd_manager->mmio)) { + dev_err(dev, "mmio not found\n"); + return PTR_ERR(amd_manager->mmio); + } + amd_manager->instance = pdata->instance; + amd_manager->mmio = amd_manager->acp_mmio + + (amd_manager->instance * SDW_MANAGER_REG_OFFSET); + amd_manager->sdw_lock = pdata->sdw_lock; + amd_manager->cols_index = sdw_find_col_index(AMD_SDW_DEFAULT_COLUMNS); + amd_manager->rows_index = sdw_find_row_index(AMD_SDW_DEFAULT_ROWS); + amd_manager->dev = dev; + amd_manager->bus.ops = &amd_sdw_ops; + amd_manager->bus.port_ops = &amd_sdw_port_ops; + amd_manager->bus.compute_params = &amd_sdw_compute_params; + amd_manager->bus.clk_stop_timeout = 200; + amd_manager->bus.link_id = amd_manager->instance; + switch (amd_manager->instance) { + case ACP_SDW0: + amd_manager->num_dout_ports = AMD_SDW0_MAX_TX_PORTS; + amd_manager->num_din_ports = AMD_SDW0_MAX_RX_PORTS; + break; + case ACP_SDW1: + amd_manager->num_dout_ports = AMD_SDW1_MAX_TX_PORTS; + amd_manager->num_din_ports = AMD_SDW1_MAX_RX_PORTS; + break; + default: + return -EINVAL; + } + amd_manager->reg_mask = &sdw_manager_reg_mask_array[amd_manager->instance]; + params = &amd_manager->bus.params; + params->max_dr_freq = AMD_SDW_DEFAULT_CLK_FREQ * 2; + params->curr_dr_freq = AMD_SDW_DEFAULT_CLK_FREQ * 2; + params->col = AMD_SDW_DEFAULT_COLUMNS; + params->row = AMD_SDW_DEFAULT_ROWS; + prop = &amd_manager->bus.prop; + prop->clk_freq = &amd_sdw_freq_tbl[0]; + prop->mclk_freq = AMD_SDW_BUS_BASE_FREQ; + + ret = sdw_bus_master_add(&amd_manager->bus, dev, dev->fwnode); + if (ret) { + dev_err(dev, "Failed to register Soundwire manager(%d)\n", ret); + return ret; + } + dev_set_drvdata(dev, amd_manager); + INIT_WORK(&amd_manager->probe_work, amd_sdw_probe_work); + schedule_work(&amd_manager->probe_work); + return 0; +} + +static int amd_sdw_manager_remove(struct platform_device *pdev) +{ + struct amd_sdw_manager *amd_manager = dev_get_drvdata(&pdev->dev); + int ret; + + if (!amd_manager) + return -ENODEV; + cancel_work_sync(&amd_manager->probe_work); + amd_disable_sdw_interrupts(amd_manager); + sdw_bus_master_delete(&amd_manager->bus); + ret = amd_disable_sdw_manager(amd_manager); + return ret; +} + +static struct platform_driver amd_sdw_driver = { + .probe = &amd_sdw_manager_probe, + .remove = &amd_sdw_manager_remove, + .driver = { + .name = "amd_sdw_manager", + } +}; +module_platform_driver(amd_sdw_driver); + +MODULE_AUTHOR("Vijendar.Mukunda(a)amd.com"); +MODULE_DESCRIPTION("AMD soundwire driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/drivers/soundwire/amd_manager.h b/drivers/soundwire/amd_manager.h new file mode 100644 index 000000000000..811ed9ee3d86 --- /dev/null +++ b/drivers/soundwire/amd_manager.h @@ -0,0 +1,252 @@ +/* SPDX-License-Identifier: GPL-2.0+ */ +/* + * Copyright (C) 2023 Advanced Micro Devices, Inc. All rights reserved. + */ + +#ifndef __AMD_MANAGER_H +#define __AMD_MANAGER_H + +#include <linux/soundwire/sdw_amd.h> + +#define SDW_MANAGER_REG_OFFSET 0xC00 +#define AMD_SDW_DEFAULT_ROWS 50 +#define AMD_SDW_DEFAULT_COLUMNS 10 +#define ACP_PAD_PULLDOWN_CTRL 0x0001448 +#define ACP_SW_PAD_KEEPER_EN 0x0001454 +#define ACP_SW0_WAKE_EN 0x0001458 +#define ACP_EXTERNAL_INTR_CNTL0 0x0001A04 +#define ACP_EXTERNAL_INTR_STAT0 0x0001A0C +#define ACP_EXTERNAL_INTR_CNTL(i) (ACP_EXTERNAL_INTR_CNTL0 + ((i) * 4)) +#define ACP_EXTERNAL_INTR_STAT(i) (ACP_EXTERNAL_INTR_STAT0 + ((i) * 4)) +#define ACP_SW_WAKE_EN(i) (ACP_SW0_WAKE_EN + ((i) * 8)) + +#define ACP_SW_EN 0x0003000 +#define ACP_SW_EN_STATUS 0x0003004 +#define ACP_SW_FRAMESIZE 0x0003008 +#define ACP_SW_SSP_COUNTER 0x000300C +#define ACP_SW_AUDIO0_TX_EN 0x0003010 +#define ACP_SW_AUDIO0_TX_EN_STATUS 0x0003014 +#define ACP_SW_AUDIO0_TX_FRAME_FORMAT 0x0003018 +#define ACP_SW_AUDIO0_TX_SAMPLEINTERVAL 0x000301C +#define ACP_SW_AUDIO0_TX_HCTRL_DP0 0x0003020 +#define ACP_SW_AUDIO0_TX_HCTRL_DP1 0x0003024 +#define ACP_SW_AUDIO0_TX_HCTRL_DP2 0x0003028 +#define ACP_SW_AUDIO0_TX_HCTRL_DP3 0x000302C +#define ACP_SW_AUDIO0_TX_OFFSET_DP0 0x0003030 +#define ACP_SW_AUDIO0_TX_OFFSET_DP1 0x0003034 +#define ACP_SW_AUDIO0_TX_OFFSET_DP2 0x0003038 +#define ACP_SW_AUDIO0_TX_OFFSET_DP3 0x000303C +#define ACP_SW_AUDIO0_TX_CHANNEL_ENABLE_DP0 0x0003040 +#define ACP_SW_AUDIO0_TX_CHANNEL_ENABLE_DP1 0x0003044 +#define ACP_SW_AUDIO0_TX_CHANNEL_ENABLE_DP2 0x0003048 +#define ACP_SW_AUDIO0_TX_CHANNEL_ENABLE_DP3 0x000304C +#define ACP_SW_AUDIO1_TX_EN 0x0003050 +#define ACP_SW_AUDIO1_TX_EN_STATUS 0x0003054 +#define ACP_SW_AUDIO1_TX_FRAME_FORMAT 0x0003058 +#define ACP_SW_AUDIO1_TX_SAMPLEINTERVAL 0x000305C +#define ACP_SW_AUDIO1_TX_HCTRL 0x0003060 +#define ACP_SW_AUDIO1_TX_OFFSET 0x0003064 +#define ACP_SW_AUDIO1_TX_CHANNEL_ENABLE_DP0 0x0003068 +#define ACP_SW_AUDIO2_TX_EN 0x000306C +#define ACP_SW_AUDIO2_TX_EN_STATUS 0x0003070 +#define ACP_SW_AUDIO2_TX_FRAME_FORMAT 0x0003074 +#define ACP_SW_AUDIO2_TX_SAMPLEINTERVAL 0x0003078 +#define ACP_SW_AUDIO2_TX_HCTRL 0x000307C +#define ACP_SW_AUDIO2_TX_OFFSET 0x0003080 +#define ACP_SW_AUDIO2_TX_CHANNEL_ENABLE_DP0 0x0003084 +#define ACP_SW_AUDIO0_RX_EN 0x0003088 +#define ACP_SW_AUDIO0_RX_EN_STATUS 0x000308C +#define ACP_SW_AUDIO0_RX_FRAME_FORMAT 0x0003090 +#define ACP_SW_AUDIO0_RX_SAMPLEINTERVAL 0x0003094 +#define ACP_SW_AUDIO0_RX_HCTRL_DP0 0x0003098 +#define ACP_SW_AUDIO0_RX_HCTRL_DP1 0x000309C +#define ACP_SW_AUDIO0_RX_HCTRL_DP2 0x0003100 +#define ACP_SW_AUDIO0_RX_HCTRL_DP3 0x0003104 +#define ACP_SW_AUDIO0_RX_OFFSET_DP0 0x0003108 +#define ACP_SW_AUDIO0_RX_OFFSET_DP1 0x000310C +#define ACP_SW_AUDIO0_RX_OFFSET_DP2 0x0003110 +#define ACP_SW_AUDIO0_RX_OFFSET_DP3 0x0003114 +#define ACP_SW_AUDIO0_RX_CHANNEL_ENABLE_DP0 0x0003118 +#define ACP_SW_AUDIO0_RX_CHANNEL_ENABLE_DP1 0x000311C +#define ACP_SW_AUDIO0_RX_CHANNEL_ENABLE_DP2 0x0003120 +#define ACP_SW_AUDIO0_RX_CHANNEL_ENABLE_DP3 0x0003124 +#define ACP_SW_AUDIO1_RX_EN 0x0003128 +#define ACP_SW_AUDIO1_RX_EN_STATUS 0x000312C +#define ACP_SW_AUDIO1_RX_FRAME_FORMAT 0x0003130 +#define ACP_SW_AUDIO1_RX_SAMPLEINTERVAL 0x0003134 +#define ACP_SW_AUDIO1_RX_HCTRL 0x0003138 +#define ACP_SW_AUDIO1_RX_OFFSET 0x000313C +#define ACP_SW_AUDIO1_RX_CHANNEL_ENABLE_DP0 0x0003140 +#define ACP_SW_AUDIO2_RX_EN 0x0003144 +#define ACP_SW_AUDIO2_RX_EN_STATUS 0x0003148 +#define ACP_SW_AUDIO2_RX_FRAME_FORMAT 0x000314C +#define ACP_SW_AUDIO2_RX_SAMPLEINTERVAL 0x0003150 +#define ACP_SW_AUDIO2_RX_HCTRL 0x0003154 +#define ACP_SW_AUDIO2_RX_OFFSET 0x0003158 +#define ACP_SW_AUDIO2_RX_CHANNEL_ENABLE_DP0 0x000315C +#define ACP_SW_BPT_PORT_EN 0x0003160 +#define ACP_SW_BPT_PORT_EN_STATUS 0x0003164 +#define ACP_SW_BPT_PORT_FRAME_FORMAT 0x0003168 +#define ACP_SW_BPT_PORT_SAMPLEINTERVAL 0x000316C +#define ACP_SW_BPT_PORT_HCTRL 0x0003170 +#define ACP_SW_BPT_PORT_OFFSET 0x0003174 +#define ACP_SW_BPT_PORT_CHANNEL_ENABLE 0x0003178 +#define ACP_SW_BPT_PORT_FIRST_BYTE_ADDR 0x000317C +#define ACP_SW_CLK_RESUME_CTRL 0x0003180 +#define ACP_SW_CLK_RESUME_DELAY_CNTR 0x0003184 +#define ACP_SW_BUS_RESET_CTRL 0x0003188 +#define ACP_SW_PRBS_ERR_STATUS 0x000318C +#define ACP_SW_IMM_CMD_UPPER_WORD 0x0003230 +#define ACP_SW_IMM_CMD_LOWER_QWORD 0x0003234 +#define ACP_SW_IMM_RESP_UPPER_WORD 0x0003238 +#define ACP_SW_IMM_RESP_LOWER_QWORD 0x000323C +#define ACP_SW_IMM_CMD_STS 0x0003240 +#define ACP_SW_BRA_BASE_ADDRESS 0x0003244 +#define ACP_SW_BRA_TRANSFER_SIZE 0x0003248 +#define ACP_SW_BRA_DMA_BUSY 0x000324C +#define ACP_SW_BRA_RESP 0x0003250 +#define ACP_SW_BRA_RESP_FRAME_ADDR 0x0003254 +#define ACP_SW_BRA_CURRENT_TRANSFER_SIZE 0x0003258 +#define ACP_SW_STATE_CHANGE_STATUS_0TO7 0x000325C +#define ACP_SW_STATE_CHANGE_STATUS_8TO11 0x0003260 +#define ACP_SW_STATE_CHANGE_STATUS_MASK_0TO7 0x0003264 +#define ACP_SW_STATE_CHANGE_STATUS_MASK_8TO11 0x0003268 +#define ACP_SW_CLK_FREQUENCY_CTRL 0x000326C +#define ACP_SW_ERROR_INTR_MASK 0x0003270 +#define ACP_SW_PHY_TEST_MODE_DATA_OFF 0x0003274 + +#define AMD_DELAY_LOOP_ITERATION 1000 +#define AMD_SDW_DEFAULT_CLK_FREQ 12000000 +#define AMD_SDW_RETRY_COUNT 1000 + +#define AMD_SDW_MCP_RESP_ACK BIT(0) +#define AMD_SDW_MCP_RESP_NACK BIT(1) +#define AMD_SDW_MCP_RESP_RDATA GENMASK(14, 7) + +#define AMD_SDW_MCP_CMD_SSP_TAG BIT(31) +#define AMD_SDW_MCP_CMD_COMMAND GENMASK(14, 12) +#define AMD_SDW_MCP_CMD_DEV_ADDR GENMASK(11, 8) +#define AMD_SDW_MCP_CMD_REG_ADDR_HIGH GENMASK(7, 0) +#define AMD_SDW_MCP_CMD_REG_ADDR_LOW GENMASK(31, 24) +#define AMD_SDW_MCP_CMD_REG_DATA GENMASK(14, 7) +#define AMD_SDW_MCP_SLAVE_STAT_0_3 GENMASK(14, 7) +#define AMD_SDW_MCP_SLAVE_STAT_4_11 GENMASK_ULL(39, 24) +#define AMD_SDW_MCP_SLAVE_STATUS_MASK GENMASK(1, 0) +#define AMD_SDW_MCP_SLAVE_STATUS_BITS GENMASK(3, 2) +#define AMD_SDW_MCP_SLAVE_STATUS_8TO_11 GENMASK_ULL(15, 0) +#define AMD_SDW_MCP_SLAVE_STATUS_VALID_MASK(x) BIT(((x) * 4)) +#define AMD_SDW_MCP_SLAVE_STAT_SHIFT_MASK(x) (((x) * 4) + 1) + +#define AMD_SDW_MASTER_SUSPEND_DELAY_MS 2000 +#define AMD_SDW_CLK_STOP_MAX_RETRY_COUNT 100 +#define AMD_SDW_QUIRK_MASK_BUS_ENABLE BIT(0) + +#define AMD_SDW_IMM_RES_VALID 1 +#define AMD_SDW_IMM_CMD_BUSY 2 +#define AMD_SDW_ENABLE 1 +#define AMD_SDW_DISABLE 0 +#define AMD_SDW_BUS_RESET_CLEAR_REQ 0 +#define AMD_SDW_BUS_RESET_REQ 1 +#define AMD_SDW_BUS_RESET_DONE 2 +#define AMD_SDW_BUS_BASE_FREQ 24000000 + +#define AMD_SDW0_EXT_INTR_MASK 0x200000 +#define AMD_SDW1_EXT_INTR_MASK 4 +#define AMD_SDW_IRQ_MASK_0TO7 0x77777777 +#define AMD_SDW_IRQ_MASK_8TO11 0x000D7777 +#define AMD_SDW_IRQ_ERROR_MASK 0xFF +#define AMD_SDW_MAX_FREQ_NUM 1 +#define AMD_SDW0_MAX_TX_PORTS 3 +#define AMD_SDW0_MAX_RX_PORTS 3 +#define AMD_SDW1_MAX_TX_PORTS 1 +#define AMD_SDW1_MAX_RX_PORTS 1 +#define AMD_SDW0_MAX_DAI 6 +#define AMD_SDW1_MAX_DAI 2 +#define AMD_SDW_SLAVE_0_ATTACHED 5 +#define AMD_SDW_SSP_COUNTER_VAL 3 + +#define AMD_DPN_FRAME_FMT_PFM GENMASK(1, 0) +#define AMD_DPN_FRAME_FMT_PDM GENMASK(3, 2) +#define AMD_DPN_FRAME_FMT_BLK_PKG_MODE BIT(4) +#define AMD_DPN_FRAME_FMT_BLK_GRP_CTRL GENMASK(6, 5) +#define AMD_DPN_FRAME_FMT_WORD_LEN GENMASK(12, 7) +#define AMD_DPN_FRAME_FMT_PCM_OR_PDM BIT(13) +#define AMD_DPN_HCTRL_HSTOP GENMASK(3, 0) +#define AMD_DPN_HCTRL_HSTART GENMASK(7, 4) +#define AMD_DPN_OFFSET_CTRL_1 GENMASK(7, 0) +#define AMD_DPN_OFFSET_CTRL_2 GENMASK(15, 8) +#define AMD_DPN_CH_EN_LCTRL GENMASK(2, 0) +#define AMD_DPN_CH_EN_CHMASK GENMASK(10, 3) +#define AMD_SDW_STAT_MAX_RETRY_COUNT 100 +#define AMD_SDW0_PAD_PULLDOWN_CTRL_ENABLE_MASK 0x7F9F +#define AMD_SDW1_PAD_PULLDOWN_CTRL_ENABLE_MASK 0x7FFA +#define AMD_SDW0_PAD_PULLDOWN_CTRL_DISABLE_MASK 0x60 +#define AMD_SDW1_PAD_PULLDOWN_CTRL_DISABLE_MASK 5 +#define AMD_SDW0_PAD_KEEPER_EN_MASK 1 +#define AMD_SDW1_PAD_KEEPER_EN_MASK 0x10 +#define AMD_SDW0_PAD_KEEPER_DISABLE_MASK 0x1E +#define AMD_SDW1_PAD_KEEPER_DISABLE_MASK 0xF + +enum amd_sdw_cmd_type { + AMD_SDW_CMD_PING = 0, + AMD_SDW_CMD_READ = 2, + AMD_SDW_CMD_WRITE = 3, +}; + +static u32 amd_sdw_freq_tbl[AMD_SDW_MAX_FREQ_NUM] = { + AMD_SDW_DEFAULT_CLK_FREQ, +}; + +struct sdw_manager_dp_reg { + u32 frame_fmt_reg; + u32 sample_int_reg; + u32 hctrl_dp0_reg; + u32 offset_reg; + u32 lane_ctrl_ch_en_reg; +}; + +static struct sdw_manager_dp_reg sdw0_manager_dp_reg[AMD_SDW0_MAX_DAI] = { + {ACP_SW_AUDIO0_TX_FRAME_FORMAT, ACP_SW_AUDIO0_TX_SAMPLEINTERVAL, ACP_SW_AUDIO0_TX_HCTRL_DP0, + ACP_SW_AUDIO0_TX_OFFSET_DP0, ACP_SW_AUDIO0_TX_CHANNEL_ENABLE_DP0}, + {ACP_SW_AUDIO1_TX_FRAME_FORMAT, ACP_SW_AUDIO1_TX_SAMPLEINTERVAL, ACP_SW_AUDIO1_TX_HCTRL, + ACP_SW_AUDIO1_TX_OFFSET, ACP_SW_AUDIO1_TX_CHANNEL_ENABLE_DP0}, + {ACP_SW_AUDIO2_TX_FRAME_FORMAT, ACP_SW_AUDIO2_TX_SAMPLEINTERVAL, ACP_SW_AUDIO2_TX_HCTRL, + ACP_SW_AUDIO2_TX_OFFSET, ACP_SW_AUDIO2_TX_CHANNEL_ENABLE_DP0}, + {ACP_SW_AUDIO0_RX_FRAME_FORMAT, ACP_SW_AUDIO0_RX_SAMPLEINTERVAL, ACP_SW_AUDIO0_RX_HCTRL_DP0, + ACP_SW_AUDIO0_RX_OFFSET_DP0, ACP_SW_AUDIO0_RX_CHANNEL_ENABLE_DP0}, + {ACP_SW_AUDIO1_RX_FRAME_FORMAT, ACP_SW_AUDIO1_RX_SAMPLEINTERVAL, ACP_SW_AUDIO1_RX_HCTRL, + ACP_SW_AUDIO1_RX_OFFSET, ACP_SW_AUDIO1_RX_CHANNEL_ENABLE_DP0}, + {ACP_SW_AUDIO2_RX_FRAME_FORMAT, ACP_SW_AUDIO2_RX_SAMPLEINTERVAL, ACP_SW_AUDIO2_RX_HCTRL, + ACP_SW_AUDIO2_RX_OFFSET, ACP_SW_AUDIO2_RX_CHANNEL_ENABLE_DP0}, +}; + +static struct sdw_manager_dp_reg sdw1_manager_dp_reg[AMD_SDW1_MAX_DAI] = { + {ACP_SW_AUDIO1_TX_FRAME_FORMAT, ACP_SW_AUDIO1_TX_SAMPLEINTERVAL, ACP_SW_AUDIO1_TX_HCTRL, + ACP_SW_AUDIO1_TX_OFFSET, ACP_SW_AUDIO1_TX_CHANNEL_ENABLE_DP0}, + {ACP_SW_AUDIO1_RX_FRAME_FORMAT, ACP_SW_AUDIO1_RX_SAMPLEINTERVAL, ACP_SW_AUDIO1_RX_HCTRL, + ACP_SW_AUDIO1_RX_OFFSET, ACP_SW_AUDIO1_RX_CHANNEL_ENABLE_DP0} +}; + +static struct sdw_manager_reg_mask sdw_manager_reg_mask_array[2] = { + { + AMD_SDW0_PAD_KEEPER_EN_MASK, + AMD_SDW0_PAD_PULLDOWN_CTRL_ENABLE_MASK, + AMD_SDW0_EXT_INTR_MASK + }, + { + AMD_SDW1_PAD_KEEPER_EN_MASK, + AMD_SDW1_PAD_PULLDOWN_CTRL_ENABLE_MASK, + AMD_SDW1_EXT_INTR_MASK + } +}; + +static inline u32 acp_reg_readl(void __iomem *base_addr) +{ + return readl(base_addr); +} + +static inline void acp_reg_writel(u32 val, void __iomem *base_addr) +{ + writel(val, base_addr); +} +#endif diff --git a/include/linux/soundwire/sdw_amd.h b/include/linux/soundwire/sdw_amd.h new file mode 100644 index 000000000000..922d30a540fd --- /dev/null +++ b/include/linux/soundwire/sdw_amd.h @@ -0,0 +1,67 @@ +/* SPDX-License-Identifier: GPL-2.0+ */ +/* + * Copyright (C) 2023 Advanced Micro Devices, Inc. All rights reserved. + */ + +#ifndef __SDW_AMD_H +#define __SDW_AMD_H + +#include <linux/soundwire/sdw.h> + +#define ACP_SDW0 0 +#define ACP_SDW1 1 + +struct acp_sdw_pdata { + u16 instance; + /* mutex to protect acp common register access */ + struct mutex *sdw_lock; +}; + +struct sdw_manager_reg_mask { + u32 sw_pad_enable_mask; + u32 sw_pad_pulldown_mask; + u32 acp_sdw_intr_mask; +}; + +/** + * struct amd_sdw_manager - amd manager driver context + * @bus: bus handle + * @dev: linux device + * @mmio: soundwire registers mmio base + * @acp_mmio: acp registers mmio base + * @reg_mask: register mask structure per manager instance + * @probe_work: soundwire manager probe workqueue + * @sdw_lock: mutex to protect acp share register access + * @num_din_ports: number of input ports + * @num_dout_ports: number of output ports + * @cols_index: Column index in frame shape + * @rows_index: Rows index in frame shape + * @instance: soundwire manager instance + * @quirks: soundwire manager quirks + * @wake_en_mask: wake enable mask per soundwire manager + * @power_mode_mask: flag interprets amd soundwire manager power mode + */ +struct amd_sdw_manager { + struct sdw_bus bus; + struct device *dev; + + void __iomem *mmio; + void __iomem *acp_mmio; + + struct sdw_manager_reg_mask *reg_mask; + struct work_struct probe_work; + /* mutex to protect acp common register access */ + struct mutex *sdw_lock; + + int num_din_ports; + int num_dout_ports; + + int cols_index; + int rows_index; + + u32 instance; + u32 quirks; + u32 wake_en_mask; + u32 power_mode_mask; +}; +#endif -- 2.34.1

3 4

[PATCH V2 5/8] soundwire: amd: add soundwire manager interrupt handling
by Vijendar Mukunda 15 Feb '23

15 Feb '23

Add support for handling soundwire manager interrupts. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Signed-off-by: Mastan Katragadda <Mastan.Katragadda(a)amd.com> --- drivers/soundwire/amd_manager.c | 132 ++++++++++++++++++++++++++++++ drivers/soundwire/amd_manager.h | 1 + include/linux/soundwire/sdw_amd.h | 7 ++ 3 files changed, 140 insertions(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 14c88b80ab6d..87f9a987d93a 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -417,6 +417,47 @@ static enum sdw_command_response amd_sdw_xfer_msg(struct sdw_bus *bus, struct sd return SDW_CMD_OK; } +static void amd_sdw_process_ping_status(u64 response, struct amd_sdw_manager *amd_manager) +{ + u64 slave_stat = 0; + u32 val = 0; + u16 dev_index; + + /* slave status response*/ + slave_stat = FIELD_GET(AMD_SDW_MCP_SLAVE_STAT_0_3, response); + slave_stat |= FIELD_GET(AMD_SDW_MCP_SLAVE_STAT_4_11, response) << 8; + + dev_dbg(amd_manager->dev, "%s: slave_stat:0x%llx\n", __func__, slave_stat); + for (dev_index = 0; dev_index <= SDW_MAX_DEVICES; ++dev_index) { + val = (slave_stat >> (dev_index * 2)) & AMD_SDW_MCP_SLAVE_STATUS_MASK; + dev_dbg(amd_manager->dev, "%s val:0x%x\n", __func__, val); + switch (val) { + case SDW_SLAVE_ATTACHED: + amd_manager->status[dev_index] = SDW_SLAVE_ATTACHED; + break; + case SDW_SLAVE_UNATTACHED: + amd_manager->status[dev_index] = SDW_SLAVE_UNATTACHED; + break; + case SDW_SLAVE_ALERT: + amd_manager->status[dev_index] = SDW_SLAVE_ALERT; + break; + default: + amd_manager->status[dev_index] = SDW_SLAVE_RESERVED; + break; + } + } +} + +static void amd_sdw_read_and_process_ping_status(struct amd_sdw_manager *amd_manager) +{ + u64 response = 0; + + mutex_lock(&amd_manager->bus.msg_lock); + response = amd_sdw_send_cmd_get_resp(amd_manager, 0, 0); + mutex_unlock(&amd_manager->bus.msg_lock); + amd_sdw_process_ping_status(response, amd_manager); +} + static u32 amd_sdw_read_ping_status(struct sdw_bus *bus) { struct amd_sdw_manager *amd_manager = to_amd_sdw(bus); @@ -817,6 +858,95 @@ static int amd_sdw_register_dais(struct amd_sdw_manager *amd_manager) dais, num_dais); } +static void amd_sdw_update_slave_status_work(struct work_struct *work) +{ + struct amd_sdw_manager *amd_manager = + container_of(work, struct amd_sdw_manager, amd_sdw_work); + int retry_count = 0; + + if (amd_manager->status[0] == SDW_SLAVE_ATTACHED) { + acp_reg_writel(0, amd_manager->mmio + ACP_SW_STATE_CHANGE_STATUS_MASK_0TO7); + acp_reg_writel(0, amd_manager->mmio + ACP_SW_STATE_CHANGE_STATUS_MASK_8TO11); + } + +update_status: + sdw_handle_slave_status(&amd_manager->bus, amd_manager->status); + if (amd_manager->status[0] == SDW_SLAVE_ATTACHED) { + if (retry_count++ < SDW_MAX_DEVICES) { + acp_reg_writel(AMD_SDW_IRQ_MASK_0TO7, amd_manager->mmio + + ACP_SW_STATE_CHANGE_STATUS_MASK_0TO7); + acp_reg_writel(AMD_SDW_IRQ_MASK_8TO11, + amd_manager->mmio + ACP_SW_STATE_CHANGE_STATUS_MASK_8TO11); + amd_sdw_read_and_process_ping_status(amd_manager); + goto update_status; + } else { + dev_err_ratelimited(amd_manager->dev, + "Device0 detected after %d iterations\n", + retry_count); + } + } +} + +static void amd_sdw_update_slave_status(u32 status_change_0to7, u32 status_change_8to11, + struct amd_sdw_manager *amd_manager) +{ + u64 slave_stat = 0; + u32 val = 0; + int dev_index; + + if (status_change_0to7 == AMD_SDW_SLAVE_0_ATTACHED) + memset(amd_manager->status, 0, sizeof(amd_manager->status)); + slave_stat = status_change_0to7; + slave_stat |= FIELD_GET(AMD_SDW_MCP_SLAVE_STATUS_8TO_11, status_change_8to11) << 32; + dev_dbg(amd_manager->dev, "%s: status_change_0to7:0x%x status_change_8to11:0x%x\n", + __func__, status_change_0to7, status_change_8to11); + if (slave_stat) { + for (dev_index = 0; dev_index <= SDW_MAX_DEVICES; ++dev_index) { + if (slave_stat & AMD_SDW_MCP_SLAVE_STATUS_VALID_MASK(dev_index)) { + val = (slave_stat >> AMD_SDW_MCP_SLAVE_STAT_SHIFT_MASK(dev_index)) & + AMD_SDW_MCP_SLAVE_STATUS_MASK; + switch (val) { + case SDW_SLAVE_ATTACHED: + amd_manager->status[dev_index] = SDW_SLAVE_ATTACHED; + break; + case SDW_SLAVE_UNATTACHED: + amd_manager->status[dev_index] = SDW_SLAVE_UNATTACHED; + break; + case SDW_SLAVE_ALERT: + amd_manager->status[dev_index] = SDW_SLAVE_ALERT; + break; + default: + amd_manager->status[dev_index] = SDW_SLAVE_RESERVED; + break; + } + } + } + } +} + +static void amd_sdw_irq_thread(struct work_struct *work) +{ + struct amd_sdw_manager *amd_manager = + container_of(work, struct amd_sdw_manager, amd_sdw_irq_thread); + u32 status_change_8to11; + u32 status_change_0to7; + + status_change_8to11 = acp_reg_readl(amd_manager->mmio + ACP_SW_STATE_CHANGE_STATUS_8TO11); + status_change_0to7 = acp_reg_readl(amd_manager->mmio + ACP_SW_STATE_CHANGE_STATUS_0TO7); + dev_dbg(amd_manager->dev, "%s [SDW%d] SDW INT: 0to7=0x%x, 8to11=0x%x\n", + __func__, amd_manager->instance, status_change_0to7, status_change_8to11); + if (status_change_8to11 & AMD_SDW_PREQ_INTR_STAT) { + amd_sdw_read_and_process_ping_status(amd_manager); + } else { + /* Check for the updated status on peripheral device */ + amd_sdw_update_slave_status(status_change_0to7, status_change_8to11, amd_manager); + } + if (status_change_8to11 || status_change_0to7) + schedule_work(&amd_manager->amd_sdw_work); + acp_reg_writel(0x00, amd_manager->mmio + ACP_SW_STATE_CHANGE_STATUS_8TO11); + acp_reg_writel(0x00, amd_manager->mmio + ACP_SW_STATE_CHANGE_STATUS_0TO7); +} + static void amd_sdw_probe_work(struct work_struct *work) { struct amd_sdw_manager *amd_manager = container_of(work, struct amd_sdw_manager, @@ -909,6 +1039,8 @@ static int amd_sdw_manager_probe(struct platform_device *pdev) return ret; } dev_set_drvdata(dev, amd_manager); + INIT_WORK(&amd_manager->amd_sdw_irq_thread, amd_sdw_irq_thread); + INIT_WORK(&amd_manager->amd_sdw_work, amd_sdw_update_slave_status_work); INIT_WORK(&amd_manager->probe_work, amd_sdw_probe_work); schedule_work(&amd_manager->probe_work); return 0; diff --git a/drivers/soundwire/amd_manager.h b/drivers/soundwire/amd_manager.h index 3e1bded1e769..5bcaf7a763bb 100644 --- a/drivers/soundwire/amd_manager.h +++ b/drivers/soundwire/amd_manager.h @@ -186,6 +186,7 @@ #define AMD_SDW1_PAD_KEEPER_EN_MASK 0x10 #define AMD_SDW0_PAD_KEEPER_DISABLE_MASK 0x1E #define AMD_SDW1_PAD_KEEPER_DISABLE_MASK 0xF +#define AMD_SDW_PREQ_INTR_STAT BIT(19) enum amd_sdw_cmd_type { AMD_SDW_CMD_PING = 0, diff --git a/include/linux/soundwire/sdw_amd.h b/include/linux/soundwire/sdw_amd.h index c978cfbc0207..003b8a197fc5 100644 --- a/include/linux/soundwire/sdw_amd.h +++ b/include/linux/soundwire/sdw_amd.h @@ -45,8 +45,11 @@ struct sdw_amd_dai_runtime { * @mmio: soundwire registers mmio base * @acp_mmio: acp registers mmio base * @reg_mask: register mask structure per manager instance + * @amd_sdw_irq_thread: soundwire manager irq workqueue + * @amd_sdw_work: peripheral status work queue * @probe_work: soundwire manager probe workqueue * @sdw_lock: mutex to protect acp share register access + * @status: peripheral devices status array * @num_din_ports: number of input ports * @num_dout_ports: number of output ports * @cols_index: Column index in frame shape @@ -65,10 +68,14 @@ struct amd_sdw_manager { void __iomem *acp_mmio; struct sdw_manager_reg_mask *reg_mask; + struct work_struct amd_sdw_irq_thread; + struct work_struct amd_sdw_work; struct work_struct probe_work; /* mutex to protect acp common register access */ struct mutex *sdw_lock; + enum sdw_slave_status status[SDW_MAX_DEVICES + 1]; + int num_din_ports; int num_dout_ports; -- 2.34.1

3 5

[PATCH V2 6/8] soundwire: amd: add runtime pm ops for AMD soundwire manager driver
by Vijendar Mukunda 14 Feb '23

14 Feb '23

Add support for runtime pm ops for AMD soundwire manager driver. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Signed-off-by: Mastan Katragadda <Mastan.Katragadda(a)amd.com> --- drivers/soundwire/amd_manager.c | 163 ++++++++++++++++++++++++++++++ drivers/soundwire/amd_manager.h | 3 + include/linux/soundwire/sdw_amd.h | 16 +++ 3 files changed, 182 insertions(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 87f9a987d93a..eced189ba6e0 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -14,6 +14,7 @@ #include <linux/slab.h> #include <linux/soundwire/sdw.h> #include <linux/soundwire/sdw_registers.h> +#include <linux/pm_runtime.h> #include <linux/wait.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -185,6 +186,15 @@ static void amd_disable_sdw_interrupts(struct amd_sdw_manager *amd_manager) acp_reg_writel(0x00, amd_manager->mmio + ACP_SW_ERROR_INTR_MASK); } +static int amd_deinit_sdw_manager(struct amd_sdw_manager *amd_manager) +{ + int ret; + + amd_disable_sdw_interrupts(amd_manager); + ret = amd_disable_sdw_manager(amd_manager); + return ret; +} + static void amd_sdw_set_frameshape(struct amd_sdw_manager *amd_manager) { u32 frame_size; @@ -1043,6 +1053,12 @@ static int amd_sdw_manager_probe(struct platform_device *pdev) INIT_WORK(&amd_manager->amd_sdw_work, amd_sdw_update_slave_status_work); INIT_WORK(&amd_manager->probe_work, amd_sdw_probe_work); schedule_work(&amd_manager->probe_work); + /* Enable runtime PM */ + pm_runtime_set_autosuspend_delay(dev, AMD_SDW_MASTER_SUSPEND_DELAY_MS); + pm_runtime_use_autosuspend(dev); + pm_runtime_mark_last_busy(dev); + pm_runtime_set_active(dev); + pm_runtime_enable(dev); return 0; } @@ -1057,14 +1073,161 @@ static int amd_sdw_manager_remove(struct platform_device *pdev) amd_disable_sdw_interrupts(amd_manager); sdw_bus_master_delete(&amd_manager->bus); ret = amd_disable_sdw_manager(amd_manager); + pm_runtime_disable(&pdev->dev); return ret; } +static int amd_sdw_clock_stop(struct amd_sdw_manager *amd_manager) +{ + u32 val; + u32 retry_count = 0; + int ret; + + ret = sdw_bus_prep_clk_stop(&amd_manager->bus); + if (ret < 0 && ret != -ENODATA) { + dev_err(amd_manager->dev, "prepare clock stop failed %d", ret); + return 0; + } + ret = sdw_bus_clk_stop(&amd_manager->bus); + if (ret < 0 && ret != -ENODATA) { + dev_err(amd_manager->dev, "bus clock stop failed %d", ret); + return 0; + } + + do { + val = acp_reg_readl(amd_manager->mmio + ACP_SW_CLK_RESUME_CTRL); + if (val & AMD_SDW_CLK_STOP_DONE) { + amd_manager->clk_stopped = true; + break; + } + } while (retry_count++ < AMD_SDW_CLK_STOP_MAX_RETRY_COUNT); + + if (!amd_manager->clk_stopped) { + dev_err(amd_manager->dev, "SDW%x clock stop failed\n", amd_manager->instance); + return 0; + } + + if (amd_manager->wake_en_mask) + acp_reg_writel(0x01, amd_manager->acp_mmio + ACP_SW_WAKE_EN(amd_manager->instance)); + + dev_dbg(amd_manager->dev, "SDW%x clock stop successful\n", amd_manager->instance); + return 0; +} + +static int amd_sdw_clock_stop_exit(struct amd_sdw_manager *amd_manager) +{ + int ret; + u32 val = 0; + u32 retry_count = 0; + + if (amd_manager->clk_stopped) { + val = acp_reg_readl(amd_manager->mmio + ACP_SW_CLK_RESUME_CTRL); + val |= AMD_SDW_CLK_RESUME_REQ; + acp_reg_writel(val, amd_manager->mmio + ACP_SW_CLK_RESUME_CTRL); + do { + val = acp_reg_readl(amd_manager->mmio + ACP_SW_CLK_RESUME_CTRL); + if (val & AMD_SDW_CLK_RESUME_DONE) + break; + usleep_range(10, 100); + } while (retry_count++ < AMD_SDW_CLK_STOP_MAX_RETRY_COUNT); + if (val & AMD_SDW_CLK_RESUME_DONE) { + acp_reg_writel(0, amd_manager->mmio + ACP_SW_CLK_RESUME_CTRL); + ret = sdw_bus_exit_clk_stop(&amd_manager->bus); + if (ret < 0) + dev_err(amd_manager->dev, "bus failed to exit clock stop %d\n", + ret); + amd_manager->clk_stopped = false; + } + } + if (amd_manager->clk_stopped) { + dev_err(amd_manager->dev, "SDW%x clock stop exit failed\n", amd_manager->instance); + return 0; + } + dev_dbg(amd_manager->dev, "SDW%x clock stop exit successful\n", amd_manager->instance); + return 0; +} + +static int __maybe_unused amd_suspend_runtime(struct device *dev) +{ + struct amd_sdw_manager *amd_manager = dev_get_drvdata(dev); + struct sdw_bus *bus = &amd_manager->bus; + int ret; + + if (bus->prop.hw_disabled) { + dev_dbg(bus->dev, "Soundwire manager %d is disabled,\n", + bus->link_id); + return 0; + } + if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { + ret = amd_sdw_clock_stop(amd_manager); + if (ret) + return ret; + } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { + ret = amd_sdw_clock_stop(amd_manager); + if (ret) + return ret; + ret = amd_deinit_sdw_manager(amd_manager); + if (ret) + return ret; + } + return 0; +} + +static int __maybe_unused amd_resume_runtime(struct device *dev) +{ + struct amd_sdw_manager *amd_manager = dev_get_drvdata(dev); + struct sdw_bus *bus = &amd_manager->bus; + int ret; + u32 val = 0; + u32 retry_count = 0; + + if (bus->prop.hw_disabled) { + dev_dbg(bus->dev, "Soundwire manager %d is disabled, ignoring\n", + bus->link_id); + return 0; + } + + if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { + ret = amd_sdw_clock_stop_exit(amd_manager); + if (ret) + return ret; + } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { + val = acp_reg_readl(amd_manager->mmio + ACP_SW_CLK_RESUME_CTRL); + if (val) { + val |= AMD_SDW_CLK_RESUME_REQ; + acp_reg_writel(val, amd_manager->mmio + ACP_SW_CLK_RESUME_CTRL); + do { + val = acp_reg_readl(amd_manager->mmio + ACP_SW_CLK_RESUME_CTRL); + if (val & AMD_SDW_CLK_RESUME_DONE) + break; + usleep_range(10, 100); + } while (retry_count++ < AMD_SDW_CLK_STOP_MAX_RETRY_COUNT); + if (val & AMD_SDW_CLK_RESUME_DONE) { + acp_reg_writel(0, amd_manager->mmio + ACP_SW_CLK_RESUME_CTRL); + amd_manager->clk_stopped = false; + } + } + sdw_clear_slave_status(bus, SDW_UNATTACH_REQUEST_MASTER_RESET); + amd_init_sdw_manager(amd_manager); + amd_enable_sdw_interrupts(amd_manager); + ret = amd_enable_sdw_manager(amd_manager); + if (ret) + return ret; + amd_sdw_set_frameshape(amd_manager); + } + return 0; +} + +static const struct dev_pm_ops amd_pm = { + SET_RUNTIME_PM_OPS(amd_suspend_runtime, amd_resume_runtime, NULL) +}; + static struct platform_driver amd_sdw_driver = { .probe = &amd_sdw_manager_probe, .remove = &amd_sdw_manager_remove, .driver = { .name = "amd_sdw_manager", + .pm = &amd_pm, } }; module_platform_driver(amd_sdw_driver); diff --git a/drivers/soundwire/amd_manager.h b/drivers/soundwire/amd_manager.h index 5bcaf7a763bb..6ec37612ae4e 100644 --- a/drivers/soundwire/amd_manager.h +++ b/drivers/soundwire/amd_manager.h @@ -187,6 +187,9 @@ #define AMD_SDW0_PAD_KEEPER_DISABLE_MASK 0x1E #define AMD_SDW1_PAD_KEEPER_DISABLE_MASK 0xF #define AMD_SDW_PREQ_INTR_STAT BIT(19) +#define AMD_SDW_CLK_STOP_DONE 1 +#define AMD_SDW_CLK_RESUME_REQ 2 +#define AMD_SDW_CLK_RESUME_DONE 3 enum amd_sdw_cmd_type { AMD_SDW_CMD_PING = 0, diff --git a/include/linux/soundwire/sdw_amd.h b/include/linux/soundwire/sdw_amd.h index 003b8a197fc5..9074c7736ff7 100644 --- a/include/linux/soundwire/sdw_amd.h +++ b/include/linux/soundwire/sdw_amd.h @@ -8,6 +8,20 @@ #include <linux/soundwire/sdw.h> +/* AMD pm_runtime quirk definitions */ + +/* + * Force the clock to stop(ClockStopMode0) when suspend callback + * is invoked. + */ +#define AMD_SDW_CLK_STOP_MODE 1 + +/* + * Stop the bus when runtime suspend/system level suspend callback + * is invoked. If set, a complete bus reset and re-enumeration will + * be performed when the bus restarts. + */ +#define AMD_SDW_POWER_OFF_MODE 2 #define ACP_SDW0 0 #define ACP_SDW1 1 @@ -57,6 +71,7 @@ struct sdw_amd_dai_runtime { * @instance: soundwire manager instance * @quirks: soundwire manager quirks * @wake_en_mask: wake enable mask per soundwire manager + * @clk_stopped: flag set to true when clock is stopped * @power_mode_mask: flag interprets amd soundwire manager power mode * @dai_runtime_array: dai runtime array */ @@ -86,6 +101,7 @@ struct amd_sdw_manager { u32 quirks; u32 wake_en_mask; u32 power_mode_mask; + bool clk_stopped; struct sdw_amd_dai_runtime **dai_runtime_array; }; -- 2.34.1

3 4

[PATCH V2 7/8] soundwire: amd: handle soundwire wake enable interrupt
by Vijendar Mukunda 14 Feb '23

14 Feb '23

Add wake enable interrupt support for both the soundwire manager instances. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Signed-off-by: Mastan Katragadda <Mastan.Katragadda(a)amd.com> --- drivers/soundwire/amd_manager.c | 10 ++++++++++ drivers/soundwire/amd_manager.h | 1 + 2 files changed, 11 insertions(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index eced189ba6e0..e53dc67f8529 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -934,6 +934,13 @@ static void amd_sdw_update_slave_status(u32 status_change_0to7, u32 status_chang } } +static void amd_sdw_process_wake_event(struct amd_sdw_manager *amd_manager) +{ + pm_request_resume(amd_manager->dev); + acp_reg_writel(0x00, amd_manager->acp_mmio + ACP_SW_WAKE_EN(amd_manager->instance)); + acp_reg_writel(0x00, amd_manager->mmio + ACP_SW_STATE_CHANGE_STATUS_8TO11); +} + static void amd_sdw_irq_thread(struct work_struct *work) { struct amd_sdw_manager *amd_manager = @@ -945,6 +952,9 @@ static void amd_sdw_irq_thread(struct work_struct *work) status_change_0to7 = acp_reg_readl(amd_manager->mmio + ACP_SW_STATE_CHANGE_STATUS_0TO7); dev_dbg(amd_manager->dev, "%s [SDW%d] SDW INT: 0to7=0x%x, 8to11=0x%x\n", __func__, amd_manager->instance, status_change_0to7, status_change_8to11); + if (status_change_8to11 & AMD_SDW_WAKE_STAT_MASK) + return amd_sdw_process_wake_event(amd_manager); + if (status_change_8to11 & AMD_SDW_PREQ_INTR_STAT) { amd_sdw_read_and_process_ping_status(amd_manager); } else { diff --git a/drivers/soundwire/amd_manager.h b/drivers/soundwire/amd_manager.h index 6ec37612ae4e..86bc6d4f48bf 100644 --- a/drivers/soundwire/amd_manager.h +++ b/drivers/soundwire/amd_manager.h @@ -190,6 +190,7 @@ #define AMD_SDW_CLK_STOP_DONE 1 #define AMD_SDW_CLK_RESUME_REQ 2 #define AMD_SDW_CLK_RESUME_DONE 3 +#define AMD_SDW_WAKE_STAT_MASK BIT(16) enum amd_sdw_cmd_type { AMD_SDW_CMD_PING = 0, -- 2.34.1

3 4

[PATCH] SoC: rt5682s: Disable jack detection interrupt during suspend
by Matthias Kaehlcke 14 Feb '23

14 Feb '23

The rt5682s driver switches its regmap to cache-only when the device suspends and back to regular mode on resume. When the jack detect interrupt fires rt5682s_irq() schedules the jack detect work. This can result in invalid reads from the regmap in cache-only mode if the work runs before the device has resumed: [ 19.672162] rt5682s 2-001a: ASoC: error at soc_component_read_no_lock on rt5682s.2-001a for register: [0x000000f0] -16 Disable the jack detection interrupt during suspend and re-enable it on resume. The driver already schedules the jack detection work on resume, so any state change during suspend is still handled. Signed-off-by: Matthias Kaehlcke <mka(a)chromium.org> --- sound/soc/codecs/rt5682s.c | 10 +++++++++- sound/soc/codecs/rt5682s.h | 1 + 2 files changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index f5e5dbc3b0f0..59117ebc5ecd 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -2895,6 +2895,9 @@ static int rt5682s_suspend(struct snd_soc_component *component) { struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); + if (rt5682s->irq) + disable_irq(rt5682s->irq); + cancel_delayed_work_sync(&rt5682s->jack_detect_work); cancel_delayed_work_sync(&rt5682s->jd_check_work); @@ -2919,6 +2922,9 @@ static int rt5682s_resume(struct snd_soc_component *component) &rt5682s->jack_detect_work, msecs_to_jiffies(0)); } + if (rt5682s->irq) + enable_irq(rt5682s->irq); + return 0; } #else @@ -3259,7 +3265,9 @@ static int rt5682s_i2c_probe(struct i2c_client *i2c) ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL, rt5682s_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5682s", rt5682s); - if (ret) + if (!ret) + rt5682s->irq = i2c->irq; + else dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); } diff --git a/sound/soc/codecs/rt5682s.h b/sound/soc/codecs/rt5682s.h index 67f86a38a1cc..caa7733b430f 100644 --- a/sound/soc/codecs/rt5682s.h +++ b/sound/soc/codecs/rt5682s.h @@ -1472,6 +1472,7 @@ struct rt5682s_priv { int pll_comb; int jack_type; + unsigned int irq; int irq_work_delay_time; int wclk_enabled; }; -- 2.39.1.519.gcb327c4b5f-goog

2 2

[PATCH] ASoC: imx-pcm-rpmsg: Remove unused variable
by Deepak R Varma 14 Feb '23

14 Feb '23

Variable ret is initialed but is never modified or used except for returning the initial value 0. The value can be directly returned instead and the variable definition can be dropped. Issue identified using returnvar.cocci Coccinelle semantic patch. Signed-off-by: Deepak R Varma <drv(a)mailo.com> --- sound/soc/fsl/imx-pcm-rpmsg.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/imx-pcm-rpmsg.c b/sound/soc/fsl/imx-pcm-rpmsg.c index 2f310994f7ee..6614b3447649 100644 --- a/sound/soc/fsl/imx-pcm-rpmsg.c +++ b/sound/soc/fsl/imx-pcm-rpmsg.c @@ -140,7 +140,6 @@ static int imx_rpmsg_pcm_hw_params(struct snd_soc_component *component, { struct rpmsg_info *info = dev_get_drvdata(component->dev); struct rpmsg_msg *msg; - int ret = 0; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { msg = &info->msg[TX_HW_PARAM]; @@ -184,7 +183,7 @@ static int imx_rpmsg_pcm_hw_params(struct snd_soc_component *component, info->send_message(msg, info); - return ret; + return 0; } static snd_pcm_uframes_t imx_rpmsg_pcm_pointer(struct snd_soc_component *component, @@ -282,7 +281,6 @@ static int imx_rpmsg_pcm_close(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct rpmsg_info *info = dev_get_drvdata(component->dev); struct rpmsg_msg *msg; - int ret = 0; /* Flush work in workqueue to make TX_CLOSE is the last message */ flush_workqueue(info->rpmsg_wq); @@ -305,7 +303,7 @@ static int imx_rpmsg_pcm_close(struct snd_soc_component *component, dev_warn(rtd->dev, "Msg is dropped!, number is %d\n", info->msg_drop_count[substream->stream]); - return ret; + return 0; } static int imx_rpmsg_pcm_prepare(struct snd_soc_component *component, -- 2.34.1

2 1

[PATCH] ASoC: codecs: wcd934x: Use min macro for comparison and assignment
by Deepak R Varma 14 Feb '23

14 Feb '23

Simplify code by using min helper macro for logical evaluation and value assignment. The change also facilitates code realignment for improved readability. Proposed change is identified using minmax.cocci Coccinelle script. Signed-off-by: Deepak R Varma <drv(a)mailo.com> --- sound/soc/codecs/wcd934x.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 28175c746b9a..783479a4d535 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -4737,13 +4737,9 @@ static u32 wcd934x_get_dmic_sample_rate(struct snd_soc_component *comp, if (dec_found && adc_mux_index <= 8) { tx_fs_reg = WCD934X_CDC_TX0_TX_PATH_CTL + (16 * adc_mux_index); tx_stream_fs = snd_soc_component_read(comp, tx_fs_reg) & 0x0F; - if (tx_stream_fs <= 4) { - if (wcd->dmic_sample_rate <= - WCD9XXX_DMIC_SAMPLE_RATE_2P4MHZ) - dmic_fs = wcd->dmic_sample_rate; - else - dmic_fs = WCD9XXX_DMIC_SAMPLE_RATE_2P4MHZ; - } else + if (tx_stream_fs <= 4) + dmic_fs = min(wcd->dmic_sample_rate, WCD9XXX_DMIC_SAMPLE_RATE_2P4MHZ); + else dmic_fs = WCD9XXX_DMIC_SAMPLE_RATE_4P8MHZ; } else { dmic_fs = wcd->dmic_sample_rate; -- 2.34.1

2 1

[PATCH 0/5] ASoC: SOF: Intel/ipc4: Support for low power playback (d0i3)
by Peter Ujfalusi 14 Feb '23

14 Feb '23

Hi, The following series will enable the the Low Power Audio (LPA) playback on Intel platforms when using IPC4. The support is closely follows how IPC3 supports similar use case. All depending patches are upstream and our CI have been testing this feature for some time without issues. Regards, Peter --- Rander Wang (5): ASoC: SOF: Introduce a new set_pm_gate() IPC PM op ASoC: SOF: Intel: hda-dsp: use set_pm_gate according to ipc version ASoC: SOF: ipc4: Wake up dsp core before sending ipc msg ASoC: SOF: Intel: Enable d0i3 work for ipc4 ASoC: SOF: Intel: hda-dsp: Set streaming flag for d0i3 sound/soc/sof/intel/cnl.c | 2 ++ sound/soc/sof/intel/hda-dsp.c | 46 ++++++++++++++++++++++++++--------- sound/soc/sof/intel/hda-ipc.c | 28 +++++++++++++++++++++ sound/soc/sof/intel/hda.h | 2 ++ sound/soc/sof/intel/mtl.c | 2 ++ sound/soc/sof/ipc3.c | 18 ++++++++++++++ sound/soc/sof/ipc4.c | 24 ++++++++++++++++++ sound/soc/sof/sof-priv.h | 2 ++ 8 files changed, 113 insertions(+), 11 deletions(-) -- 2.39.1

2 6

[PATCH] ASoC: tlv320adcx140: extend list of supported samplerates
by Alexandru Ardelean 14 Feb '23

14 Feb '23

From: Steffen Aschbacher <steffen.aschbacher(a)stihl.de> The 'tlv320adcx140' driver currently supports 3 devices: TLV320ADC3140, TLV320ADC5140 and TLV320ADC6140. All 3 devices, support higher samplerates, up to 768-kHz according to their datasheets. In our applications, we only tested (and worked) with 96 kHz and 192 kHz. This change extends the list of supported sample-rates for these devices with 96 & 192 kHz. References: https://www.ti.com/product/TLV320ADC3140 https://www.ti.com/product/TLV320ADC5140 https://www.ti.com/product/TLV320ADC6140 Signed-off-by: Steffen Aschbacher <steffen.aschbacher(a)stihl.de> Signed-off-by: Alexandru Ardelean <alex(a)shruggie.ro> --- sound/soc/codecs/tlv320adcx140.h | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320adcx140.h b/sound/soc/codecs/tlv320adcx140.h index fd80fac8b327..27a1f1012fe2 100644 --- a/sound/soc/codecs/tlv320adcx140.h +++ b/sound/soc/codecs/tlv320adcx140.h @@ -6,7 +6,9 @@ #define _TLV320ADCX140_H #define ADCX140_RATES (SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000) + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_192000) #define ADCX140_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S20_3LE | \ -- 2.34.1

2 1

[PATCH 00/35] Documentation: correct lots of spelling errors (series 1)
by Randy Dunlap 14 Feb '23

14 Feb '23

Correct many spelling errors in Documentation/ as reported by codespell. Maintainers of specific kernel subsystems are only Cc-ed on their respective patches, not the entire series. [if all goes well] These patches are based on linux-next-20230125. [PATCH 01/35] Documentation: arm64: correct spelling [PATCH 02/35] Documentation: arm: correct spelling [PATCH 03/35] Documentation: block: correct spelling [PATCH 04/35] Documentation: bpf: correct spelling [PATCH 05/35] Documentation: core-api: correct spelling [PATCH 06/35] Documentation: fault-injection: correct spelling [PATCH 07/35] Documentation: fb: correct spelling [PATCH 08/35] Documentation: features: correct spelling [PATCH 09/35] Documentation: firmware-guide/acpi: correct spelling [PATCH 10/35] Documentation: hid: correct spelling [PATCH 11/35] Documentation: i2c: correct spelling [PATCH 12/35] Documentation: input: correct spelling [PATCH 13/35] Documentation: isdn: correct spelling [PATCH 14/35] Documentation: leds: correct spelling [PATCH 15/35] Documentation: litmus-tests: correct spelling [PATCH 16/35] Documentation: livepatch: correct spelling [PATCH 17/35] Documentation: locking: correct spelling [PATCH 18/35] Documentation: mm: correct spelling [PATCH 19/35] Documentation: openrisc: correct spelling [PATCH 20/35] Documentation: PCI: correct spelling [PATCH 21/35] Documentation: powerpc: correct spelling [PATCH 22/35] Documentation: power: correct spelling [PATCH 23/35] Documentation: s390: correct spelling [PATCH 24/35] Documentation: scheduler: correct spelling [PATCH 25/35] Documentation: security: correct spelling [PATCH 26/35] Documentation: sound: correct spelling [PATCH 27/35] Documentation: spi: correct spelling [PATCH 28/35] Documentation: target: correct spelling [PATCH 29/35] Documentation: timers: correct spelling [PATCH 30/35] Documentation: tools/rtla: correct spelling [PATCH 31/35] Documentation: trace: correct spelling [PATCH 32/35] Documentation: usb: correct spelling [PATCH 33/35] Documentation: w1: correct spelling [PATCH 34/35] Documentation: x86: correct spelling [PATCH 35/35] Documentation: xtensa: correct spelling Documentation/PCI/endpoint/pci-vntb-howto.rst | 2 - Documentation/PCI/msi-howto.rst | 2 - Documentation/arm/arm.rst | 2 - Documentation/arm/ixp4xx.rst | 4 +- Documentation/arm/keystone/knav-qmss.rst | 2 - Documentation/arm/stm32/stm32-dma-mdma-chaining.rst | 6 +-- Documentation/arm/sunxi/clocks.rst | 2 - Documentation/arm/swp_emulation.rst | 2 - Documentation/arm/tcm.rst | 2 - Documentation/arm/vlocks.rst | 2 - Documentation/arm64/booting.rst | 2 - Documentation/arm64/elf_hwcaps.rst | 2 - Documentation/arm64/sve.rst | 4 +- Documentation/block/data-integrity.rst | 2 - Documentation/bpf/libbpf/libbpf_naming_convention.rst | 6 +-- Documentation/bpf/map_xskmap.rst | 2 - Documentation/bpf/ringbuf.rst | 4 +- Documentation/bpf/verifier.rst | 2 - Documentation/core-api/packing.rst | 2 - Documentation/core-api/padata.rst | 2 - Documentation/fault-injection/fault-injection.rst | 2 - Documentation/fb/sm712fb.rst | 2 - Documentation/fb/sstfb.rst | 2 - Documentation/features/core/thread-info-in-task/arch-support.txt | 2 - Documentation/firmware-guide/acpi/acpi-lid.rst | 2 - Documentation/firmware-guide/acpi/namespace.rst | 2 - Documentation/hid/hid-alps.rst | 2 - Documentation/hid/hid-bpf.rst | 2 - Documentation/hid/hiddev.rst | 2 - Documentation/hid/hidraw.rst | 2 - Documentation/hid/intel-ish-hid.rst | 2 - Documentation/i2c/gpio-fault-injection.rst | 2 - Documentation/i2c/smbus-protocol.rst | 2 - Documentation/input/devices/iforce-protocol.rst | 2 - Documentation/input/multi-touch-protocol.rst | 2 - Documentation/isdn/interface_capi.rst | 2 - Documentation/isdn/m_isdn.rst | 2 - Documentation/leds/leds-qcom-lpg.rst | 4 +- Documentation/litmus-tests/README | 2 - Documentation/livepatch/reliable-stacktrace.rst | 2 - Documentation/locking/lockdep-design.rst | 4 +- Documentation/locking/locktorture.rst | 2 - Documentation/locking/locktypes.rst | 2 - Documentation/locking/preempt-locking.rst | 2 - Documentation/mm/hmm.rst | 4 +- Documentation/mm/hwpoison.rst | 2 - Documentation/openrisc/openrisc_port.rst | 4 +- Documentation/power/suspend-and-interrupts.rst | 2 - Documentation/powerpc/kasan.txt | 2 - Documentation/powerpc/papr_hcalls.rst | 2 - Documentation/powerpc/qe_firmware.rst | 4 +- Documentation/powerpc/vas-api.rst | 4 +- Documentation/s390/pci.rst | 4 +- Documentation/s390/vfio-ccw.rst | 2 - Documentation/scheduler/sched-bwc.rst | 2 - Documentation/scheduler/sched-energy.rst | 4 +- Documentation/security/digsig.rst | 4 +- Documentation/security/keys/core.rst | 2 - Documentation/security/secrets/coco.rst | 2 - Documentation/sound/alsa-configuration.rst | 8 ++-- Documentation/sound/cards/audigy-mixer.rst | 2 - Documentation/sound/cards/maya44.rst | 2 - Documentation/sound/cards/sb-live-mixer.rst | 2 - Documentation/sound/designs/jack-controls.rst | 2 - Documentation/sound/designs/seq-oss.rst | 2 - Documentation/sound/hd-audio/notes.rst | 2 - Documentation/spi/pxa2xx.rst | 12 +++--- Documentation/spi/spi-lm70llp.rst | 2 - Documentation/spi/spi-summary.rst | 2 - Documentation/target/tcmu-design.rst | 2 - Documentation/timers/hrtimers.rst | 2 - Documentation/tools/rtla/rtla-timerlat-top.rst | 2 - Documentation/trace/coresight/coresight-etm4x-reference.rst | 2 - Documentation/trace/events.rst | 6 +-- Documentation/trace/fprobe.rst | 2 - Documentation/trace/ftrace-uses.rst | 2 - Documentation/trace/hwlat_detector.rst | 2 - Documentation/trace/rv/runtime-verification.rst | 2 - Documentation/trace/uprobetracer.rst | 2 - Documentation/usb/chipidea.rst | 19 +++++----- Documentation/usb/gadget-testing.rst | 2 - Documentation/usb/mass-storage.rst | 2 - Documentation/w1/w1-netlink.rst | 2 - Documentation/x86/boot.rst | 2 - Documentation/x86/buslock.rst | 2 - Documentation/x86/mds.rst | 2 - Documentation/x86/resctrl.rst | 2 - Documentation/x86/sgx.rst | 2 - Documentation/xtensa/atomctl.rst | 2 - 89 files changed, 124 insertions(+), 123 deletions(-) Cc: Jonathan Corbet <corbet(a)lwn.net> Cc: Catalin Marinas <catalin.marinas(a)arm.com> Cc: Will Deacon <will(a)kernel.org> Cc: Russell King <linux(a)armlinux.org.uk> Cc: Jens Axboe <axboe(a)kernel.dk> Cc: Andrii Nakryiko <andrii(a)kernel.org> Cc: Alexei Starovoitov <ast(a)kernel.org> Cc: Daniel Borkmann <daniel(a)iogearbox.net> Cc: Vladimir Oltean <olteanv(a)gmail.com> Cc: Steffen Klassert <steffen.klassert(a)secunet.com> Cc: Daniel Jordan <daniel.m.jordan(a)oracle.com> Cc: Akinobu Mita <akinobu.mita(a)gmail.com> Cc: Helge Deller <deller(a)gmx.de> Cc: "Rafael J. Wysocki" <rafael(a)kernel.org> Cc: Jiri Kosina <jikos(a)kernel.org> Cc: Benjamin Tissoires <benjamin.tissoires(a)redhat.com> Cc: Srinivas Pandruvada <srinivas.pandruvada(a)linux.intel.com> Cc: Wolfram Sang <wsa(a)kernel.org> Cc: Dmitry Torokhov <dmitry.torokhov(a)gmail.com> Cc: Henrik Rydberg <rydberg(a)bitmath.org> Cc: Karsten Keil <isdn(a)linux-pingi.de> Cc: Pavel Machek <pavel(a)ucw.cz> Cc: Lee Jones <lee(a)kernel.org> Cc: Josh Poimboeuf <jpoimboe(a)kernel.org> Cc: Miroslav Benes <mbenes(a)suse.cz> Cc: Petr Mladek <pmladek(a)suse.com> Cc: Peter Zijlstra <peterz(a)infradead.org> Cc: Ingo Molnar <mingo(a)redhat.com> Cc: Jérôme Glisse <jglisse(a)redhat.com> Cc: Naoya Horiguchi <naoya.horiguchi(a)nec.com> Cc: Miaohe Lin <linmiaohe(a)huawei.com> Cc: Jonas Bonn <jonas(a)southpole.se> Cc: Stefan Kristiansson <stefan.kristiansson(a)saunalahti.fi> Cc: Stafford Horne <shorne(a)gmail.com> Cc: Bjorn Helgaas <bhelgaas(a)google.com> Cc: Lorenzo Pieralisi <lpieralisi(a)kernel.org> Cc: Marc Zyngier <maz(a)kernel.org> Cc: Michael Ellerman <mpe(a)ellerman.id.au> Cc: Len Brown <len.brown(a)intel.com> Cc: Heiko Carstens <hca(a)linux.ibm.com> Cc: Vasily Gorbik <gor(a)linux.ibm.com> Cc: Alexander Gordeev <agordeev(a)linux.ibm.com> Cc: Juri Lelli <juri.lelli(a)redhat.com> Cc: Vincent Guittot <vincent.guittot(a)linaro.org> Cc: David Howells <dhowells(a)redhat.com> Cc: Jarkko Sakkinen <jarkko(a)kernel.org> Cc: Paul Moore <paul(a)paul-moore.com> Cc: James Morris <jmorris(a)namei.org> Cc: "Serge E. Hallyn" <serge(a)hallyn.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: "Martin K. Petersen" <martin.petersen(a)oracle.com> Cc: Daniel Bristot de Oliveira <bristot(a)kernel.org> Cc: Steven Rostedt <rostedt(a)goodmis.org> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Masami Hiramatsu <mhiramat(a)kernel.org> Cc: Mathieu Poirier <mathieu.poirier(a)linaro.org> Cc: Suzuki K Poulose <suzuki.poulose(a)arm.com> Cc: Evgeniy Polyakov <zbr(a)ioremap.net> Cc: Fenghua Yu <fenghua.yu(a)intel.com> Cc: Reinette Chatre <reinette.chatre(a)intel.com> Cc: Thomas Gleixner <tglx(a)linutronix.de> Cc: Borislav Petkov <bp(a)alien8.de> Cc: Chris Zankel <chris(a)zankel.net> Cc: Max Filippov <jcmvbkbc(a)gmail.com> Cc: alsa-devel(a)alsa-project.org Cc: coresight(a)lists.linaro.org Cc: bpf(a)vger.kernel.org Cc: dri-devel(a)lists.freedesktop.org Cc: isdn4linux(a)listserv.isdn4linux.de Cc: keyrings(a)vger.kernel.org Cc: linux-acpi(a)vger.kernel.org Cc: linux-block(a)vger.kernel.org Cc: linux-crypto(a)vger.kernel.org Cc: linux-doc(a)vger.kernel.org Cc: linux-fbdev(a)vger.kernel.org Cc: linux-i2c(a)vger.kernel.org Cc: linux-input(a)vger.kernel.org Cc: linux-leds(a)vger.kernel.org Cc: linux-pci(a)vger.kernel.org Cc: linux-s390(a)vger.kernel.org Cc: linux-scsi(a)vger.kernel.org Cc: linux-sgx(a)vger.kernel.org Cc: linux-spi(a)vger.kernel.org Cc: linux-trace-devel(a)vger.kernel.org Cc: linux-trace-kernel(a)vger.kernel.org Cc: live-patching(a)vger.kernel.org Cc: linux-pm(a)vger.kernel.org Cc: linux-security-module(a)vger.kernel.org Cc: linux-usb(a)vger.kernel.org Cc: netdev(a)vger.kernel.org Cc: target-devel(a)vger.kernel.org Cc: linux-mm(a)kvack.org Cc: openrisc(a)lists.librecores.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-xtensa(a)linux-xtensa.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: x86(a)kernel.org

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[PATCH 1/2] ALSA: hda/realtek: fix mute/micmute LEDs don't work for a HP platform.
by Andy Chi 14 Feb '23

14 Feb '23

There is a HP platform needs ALC236_FIXUP_HP_GPIO_LED quirk to make mic-mute/audio-mute working. Signed-off-by: Andy Chi <andy.chi(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1134a493d225..7b9fb38ff732 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9436,6 +9436,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8b5e, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8b7a, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b7d, "HP", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8b87, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b8a, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b8b, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b8d, "HP", ALC236_FIXUP_HP_GPIO_LED), -- 2.34.1

2 9

[PATCH] sound: soc: codec: tlv320adcx140: fix 'ti,gpio-config' DT property init
by Alexandru Ardelean 14 Feb '23

14 Feb '23

From: Steffen Aschbacher <steffen.aschbacher(a)stihl.de> When the 'ti,gpio-config' property is not defined, the device_property_count_u32() will return an error, rather than zero. The current check, only handles a return value of zero, which assumes that the property is defined and has no elements. This change extends the check to also handle for an error case (most likely to be hit by the case that the 'ti,gpio-config' is not defined). Semantically, this would be the same as 'device_property_count_u32()' returning zero. In case that 'ti,gpio-config' is defined, and returns a 'gpio_count' value that is incorrect, there is an 'if (gpio_count != ADCX140_NUM_GPIO_CFGS)' check, a few lines lower that will return -EINVAL. This means that someone tried to define 'ti,gpio-config', but with the wrong number of GPIOs. Signed-off-by: Steffen Aschbacher <steffen.aschbacher(a)stihl.de> Signed-off-by: Alexandru Ardelean <alex(a)shruggie.ro> --- sound/soc/codecs/tlv320adcx140.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 91a22d927915..530f321d08e9 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -925,7 +925,7 @@ static int adcx140_configure_gpio(struct adcx140_priv *adcx140) gpio_count = device_property_count_u32(adcx140->dev, "ti,gpio-config"); - if (gpio_count == 0) + if (gpio_count <= 0) return 0; if (gpio_count != ADCX140_NUM_GPIO_CFGS) -- 2.34.1

2 4

[PATCH V2 3/8] soundwire: amd: register soundwire manager dai ops
by Vijendar Mukunda 14 Feb '23

14 Feb '23

Register dai ops for soundwire manager instances. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 182 ++++++++++++++++++++++++++++++ drivers/soundwire/amd_manager.h | 18 +++ include/linux/soundwire/sdw_amd.h | 18 +++ 3 files changed, 218 insertions(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index cd1e5a3d5995..14c88b80ab6d 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -641,6 +641,182 @@ static const struct sdw_master_ops amd_sdw_ops = { .read_ping_status = amd_sdw_read_ping_status, }; +static int amd_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct amd_sdw_manager *amd_manager = snd_soc_dai_get_drvdata(dai); + struct sdw_amd_dai_runtime *dai_runtime; + struct sdw_stream_config sconfig; + struct sdw_port_config *pconfig; + int ch, dir; + int ret; + + dai_runtime = amd_manager->dai_runtime_array[dai->id]; + if (!dai_runtime) + return -EIO; + + ch = params_channels(params); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + dir = SDW_DATA_DIR_RX; + else + dir = SDW_DATA_DIR_TX; + dev_dbg(amd_manager->dev, "%s: dir:%d dai->id:0x%x\n", __func__, dir, dai->id); + + sconfig.direction = dir; + sconfig.ch_count = ch; + sconfig.frame_rate = params_rate(params); + sconfig.type = dai_runtime->stream_type; + + sconfig.bps = snd_pcm_format_width(params_format(params)); + + /* Port configuration */ + pconfig = kzalloc(sizeof(*pconfig), GFP_KERNEL); + if (!pconfig) { + ret = -ENOMEM; + goto error; + } + + pconfig->num = dai->id; + pconfig->ch_mask = (1 << ch) - 1; + ret = sdw_stream_add_master(&amd_manager->bus, &sconfig, + pconfig, 1, dai_runtime->stream); + if (ret) + dev_err(amd_manager->dev, "add manager to stream failed:%d\n", ret); + + kfree(pconfig); +error: + return ret; +} + +static int amd_sdw_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) +{ + struct amd_sdw_manager *amd_manager = snd_soc_dai_get_drvdata(dai); + struct sdw_amd_dai_runtime *dai_runtime; + int ret; + + dai_runtime = amd_manager->dai_runtime_array[dai->id]; + if (!dai_runtime) + return -EIO; + + ret = sdw_stream_remove_master(&amd_manager->bus, dai_runtime->stream); + if (ret < 0) + dev_err(dai->dev, "remove manager from stream %s failed: %d\n", + dai_runtime->stream->name, ret); + return ret; +} + +static int amd_set_sdw_stream(struct snd_soc_dai *dai, void *stream, int direction) +{ + struct amd_sdw_manager *amd_manager = snd_soc_dai_get_drvdata(dai); + struct sdw_amd_dai_runtime *dai_runtime; + + dai_runtime = amd_manager->dai_runtime_array[dai->id]; + if (stream) { + /* first paranoia check */ + if (dai_runtime) { + dev_err(dai->dev, + "dai_runtime already allocated for dai %s\n", + dai->name); + return -EINVAL; + } + + /* allocate and set dai_runtime info */ + dai_runtime = kzalloc(sizeof(*dai_runtime), GFP_KERNEL); + if (!dai_runtime) + return -ENOMEM; + + dai_runtime->stream_type = SDW_STREAM_PCM; + dai_runtime->bus = &amd_manager->bus; + dai_runtime->stream = stream; + amd_manager->dai_runtime_array[dai->id] = dai_runtime; + } else { + /* second paranoia check */ + if (!dai_runtime) { + dev_err(dai->dev, + "dai_runtime not allocated for dai %s\n", + dai->name); + return -EINVAL; + } + + /* for NULL stream we release allocated dai_runtime */ + kfree(dai_runtime); + amd_manager->dai_runtime_array[dai->id] = NULL; + } + return 0; +} + +static int amd_pcm_set_sdw_stream(struct snd_soc_dai *dai, void *stream, int direction) +{ + return amd_set_sdw_stream(dai, stream, direction); +} + +static void *amd_get_sdw_stream(struct snd_soc_dai *dai, int direction) +{ + struct amd_sdw_manager *amd_manager = snd_soc_dai_get_drvdata(dai); + struct sdw_amd_dai_runtime *dai_runtime; + + dai_runtime = amd_manager->dai_runtime_array[dai->id]; + if (!dai_runtime) + return ERR_PTR(-EINVAL); + + return dai_runtime->stream; +} + +static const struct snd_soc_dai_ops amd_sdw_dai_ops = { + .hw_params = amd_sdw_hw_params, + .hw_free = amd_sdw_hw_free, + .set_stream = amd_pcm_set_sdw_stream, + .get_stream = amd_get_sdw_stream, +}; + +static const struct snd_soc_component_driver amd_sdw_dai_component = { + .name = "soundwire", +}; + +static int amd_sdw_register_dais(struct amd_sdw_manager *amd_manager) +{ + struct sdw_amd_dai_runtime **dai_runtime_array; + struct snd_soc_dai_driver *dais; + struct snd_soc_pcm_stream *stream; + struct device *dev; + int i, num_dais; + + dev = amd_manager->dev; + num_dais = amd_manager->num_dout_ports + amd_manager->num_din_ports; + dais = devm_kcalloc(dev, num_dais, sizeof(*dais), GFP_KERNEL); + if (!dais) + return -ENOMEM; + + dai_runtime_array = devm_kcalloc(dev, num_dais, + sizeof(struct sdw_amd_dai_runtime *), + GFP_KERNEL); + if (!dai_runtime_array) + return -ENOMEM; + amd_manager->dai_runtime_array = dai_runtime_array; + for (i = 0; i < num_dais; i++) { + dais[i].name = devm_kasprintf(dev, GFP_KERNEL, "SDW%d Pin%d", amd_manager->instance, + i); + if (!dais[i].name) + return -ENOMEM; + if (i < amd_manager->num_dout_ports) + stream = &dais[i].playback; + else + stream = &dais[i].capture; + + stream->channels_min = 2; + stream->channels_max = 2; + stream->rates = SNDRV_PCM_RATE_48000; + stream->formats = SNDRV_PCM_FMTBIT_S16_LE; + + dais[i].ops = &amd_sdw_dai_ops; + dais[i].id = i; + } + + return devm_snd_soc_register_component(dev, &amd_sdw_dai_component, + dais, num_dais); +} + static void amd_sdw_probe_work(struct work_struct *work) { struct amd_sdw_manager *amd_manager = container_of(work, struct amd_sdw_manager, @@ -726,6 +902,12 @@ static int amd_sdw_manager_probe(struct platform_device *pdev) dev_err(dev, "Failed to register Soundwire manager(%d)\n", ret); return ret; } + ret = amd_sdw_register_dais(amd_manager); + if (ret) { + dev_err(dev, "CPU DAI registration failed\n"); + sdw_bus_master_delete(&amd_manager->bus); + return ret; + } dev_set_drvdata(dev, amd_manager); INIT_WORK(&amd_manager->probe_work, amd_sdw_probe_work); schedule_work(&amd_manager->probe_work); diff --git a/drivers/soundwire/amd_manager.h b/drivers/soundwire/amd_manager.h index 811ed9ee3d86..3e1bded1e769 100644 --- a/drivers/soundwire/amd_manager.h +++ b/drivers/soundwire/amd_manager.h @@ -205,6 +205,24 @@ struct sdw_manager_dp_reg { u32 lane_ctrl_ch_en_reg; }; +/* + * SDW0 Manager instance registers 6 CPU DAI (3 TX & 3 RX Ports) + * whereas SDW1 Manager Instance registers 2 CPU DAI (one TX & one RX port) + * Below is the CPU DAI <->Manager port number mapping + * i.e SDW0 Pin0 -> port number 0 -> AUDIO0 TX + * SDW0 Pin1 -> Port number 1 -> AUDIO1 TX + * SDW0 Pin2 -> Port number 2 -> AUDIO2 TX + * SDW0 Pin3 -> port number 3 -> AUDIO0 RX + * SDW0 Pin4 -> Port number 4 -> AUDIO1 RX + * SDW0 Pin5 -> Port number 5 -> AUDIO2 RX + * Whereas for SDW1 instance + * SDW1 Pin0 -> port number 0 -> AUDIO1 TX + * SDW1 Pin1 -> Port number 1 -> AUDIO1 RX + * Same mapping should be used for programming DMA controller registers in Soundwire DMA driver. + * i.e if AUDIO0 TX channel is selected then we need to use AUDIO0 TX registers for DMA programming + * in Soundwire DMA driver. + */ + static struct sdw_manager_dp_reg sdw0_manager_dp_reg[AMD_SDW0_MAX_DAI] = { {ACP_SW_AUDIO0_TX_FRAME_FORMAT, ACP_SW_AUDIO0_TX_SAMPLEINTERVAL, ACP_SW_AUDIO0_TX_HCTRL_DP0, ACP_SW_AUDIO0_TX_OFFSET_DP0, ACP_SW_AUDIO0_TX_CHANNEL_ENABLE_DP0}, diff --git a/include/linux/soundwire/sdw_amd.h b/include/linux/soundwire/sdw_amd.h index 922d30a540fd..c978cfbc0207 100644 --- a/include/linux/soundwire/sdw_amd.h +++ b/include/linux/soundwire/sdw_amd.h @@ -23,6 +23,21 @@ struct sdw_manager_reg_mask { u32 acp_sdw_intr_mask; }; +/** + * struct sdw_amd_dai_runtime: AMD sdw dai runtime data + * + * @name: SoundWire stream name + * @stream: stream runtime + * @bus: Bus handle + * @stream_type: Stream type + */ +struct sdw_amd_dai_runtime { + char *name; + struct sdw_stream_runtime *stream; + struct sdw_bus *bus; + enum sdw_stream_type stream_type; +}; + /** * struct amd_sdw_manager - amd manager driver context * @bus: bus handle @@ -40,6 +55,7 @@ struct sdw_manager_reg_mask { * @quirks: soundwire manager quirks * @wake_en_mask: wake enable mask per soundwire manager * @power_mode_mask: flag interprets amd soundwire manager power mode + * @dai_runtime_array: dai runtime array */ struct amd_sdw_manager { struct sdw_bus bus; @@ -63,5 +79,7 @@ struct amd_sdw_manager { u32 quirks; u32 wake_en_mask; u32 power_mode_mask; + + struct sdw_amd_dai_runtime **dai_runtime_array; }; #endif -- 2.34.1

3 2

remove arch/sh
by Christoph Hellwig 13 Feb '23

13 Feb '23

18 96

[PATCH 0/3][next] ASoC: codecs: Modify error implicit declaration of function and misspelling
by wangweidong.a＠awinic.com 13 Feb '23

13 Feb '23

From: Weidong Wang <wangweidong.a(a)awinic.com> Change the function name aw88395_plack_event to aw88395_playback_event Change hdrlen to hdr_len Add select GPIOLIB to the sound/soc/codecs/Kconfig file Weidong Wang (3): ASoC: codecs: Fixed a spelling error in the function name ASoC: codecs: Modify the log print statement ASoC: codecs: Modify error: implicit declaration of function 'gpiod_set_value_cansleep' sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/aw88395/aw88395.c | 4 ++-- sound/soc/codecs/aw88395/aw88395_lib.c | 2 +- 3 files changed, 4 insertions(+), 3 deletions(-) base-commit: 6ba8a227fd19d19779005fb66ad7562608e1df83 -- 2.39.1

2 4

[PATCH -next] ASoC: codecs: Remove unneeded semicolon
by Yang Li 13 Feb '23

13 Feb '23

./sound/soc/codecs/peb2466.c:1851:2-3: Unneeded semicolon ./sound/soc/codecs/peb2466.c:1887:2-3: Unneeded semicolon Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Link: https://bugzilla.openanolis.cn/show_bug.cgi?id=4045 Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- sound/soc/codecs/peb2466.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/peb2466.c b/sound/soc/codecs/peb2466.c index 4640b1860312..5dec69be0acb 100644 --- a/sound/soc/codecs/peb2466.c +++ b/sound/soc/codecs/peb2466.c @@ -1848,7 +1848,7 @@ static int peb2466_chip_direction_input(struct gpio_chip *c, unsigned int offset if (offset < 16) { /* SOx_{0,1} */ return -EINVAL; - }; + } ret = peb2466_chip_gpio_offset_to_dir_regmask(offset, &xr_reg, &mask); if (ret) { @@ -1884,7 +1884,7 @@ static int peb2466_chip_direction_output(struct gpio_chip *c, unsigned int offse if (offset < 16) { /* SOx_{0,1} */ return 0; - }; + } ret = peb2466_chip_gpio_offset_to_dir_regmask(offset, &xr_reg, &mask); if (ret) { -- 2.20.1.7.g153144c

3 2

[PATCH] ASoC: ep93xx: ignore 0 Hz sysclk
by Alexander Sverdlin 13 Feb '23

13 Feb '23

Commit 2458adb8f92a ("SoC: simple-card-utils: set 0Hz to sysclk when shutdown") added a call to snd_soc_dai_set_sysclk() with 0 Hz frequency. Being propagated further it causes a division by zero in clk-ep93xx driver: Division by zero in kernel. CPU: 0 PID: 52 Comm: aplay Tainted: G W 6.2.0-rc4-... #1 Hardware name: Generic DT based system unwind_backtrace from show_stack+0x10/0x18 show_stack from dump_stack_lvl+0x28/0x34 dump_stack_lvl from __div0+0x10/0x1c __div0 from Ldiv0+0x8/0x1c Ldiv0 from ep93xx_mux_determine_rate+0x78/0x1d0 ep93xx_mux_determine_rate from clk_core_round_rate_nolock+0x48/0xc8 clk_core_round_rate_nolock from clk_core_set_rate_nolock+0x48/0x160 clk_core_set_rate_nolock from clk_set_rate+0x30/0x8c clk_set_rate from ep93xx_i2s_set_sysclk+0x30/0x6c ep93xx_i2s_set_sysclk from snd_soc_dai_set_sysclk+0x3c/0xa4 snd_soc_dai_set_sysclk from asoc_simple_shutdown+0xb8/0x164 asoc_simple_shutdown from snd_soc_link_shutdown+0x44/0x54 snd_soc_link_shutdown from soc_pcm_clean+0x78/0x180 soc_pcm_clean from soc_pcm_close+0x28/0x40 soc_pcm_close from snd_pcm_release_substream.part.0+0x3c/0x84 snd_pcm_release_substream.part.0 from snd_pcm_release+0x40/0x88 snd_pcm_release from __fput+0x74/0x278 There has been commit f1879d7b98dc ("ASoC: rockchip: ignore 0Hz sysclk"), but it prepared by far not all drivers. Signed-off-by: Alexander Sverdlin <alexander.sverdlin(a)gmail.com> --- sound/soc/cirrus/ep93xx-i2s.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index 982151330c89..f41712df7994 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -359,6 +359,8 @@ static int ep93xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, if (dir == SND_SOC_CLOCK_IN || clk_id != 0) return -EINVAL; + if (!freq) + return 0; return clk_set_rate(info->mclk, freq); } -- 2.39.1

2 1

[PATCH] ASoC: dt-bindings: maxim,max98095: Convert to dtschema
by Krzysztof Kozlowski 13 Feb '23

13 Feb '23

Convert the Maxim Integrated MAX98095 audio codec bindings to DT schema. Add missing sound-dai-cells during conversion. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- .../devicetree/bindings/sound/max98095.txt | 22 -------- .../bindings/sound/maxim,max98095.yaml | 54 +++++++++++++++++++ 2 files changed, 54 insertions(+), 22 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/max98095.txt create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98095.yaml diff --git a/Documentation/devicetree/bindings/sound/max98095.txt b/Documentation/devicetree/bindings/sound/max98095.txt deleted file mode 100644 index 318a4c82f17f..000000000000 --- a/Documentation/devicetree/bindings/sound/max98095.txt +++ /dev/null @@ -1,22 +0,0 @@ -MAX98095 audio CODEC - -This device supports I2C only. - -Required properties: - -- compatible : "maxim,max98095". - -- reg : The I2C address of the device. - -Optional properties: - -- clocks: The phandle of the master clock to the CODEC - -- clock-names: Should be "mclk" - -Example: - -max98095: codec@11 { - compatible = "maxim,max98095"; - reg = <0x11>; -}; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98095.yaml b/Documentation/devicetree/bindings/sound/maxim,max98095.yaml new file mode 100644 index 000000000000..77544a9e1587 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98095.yaml @@ -0,0 +1,54 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98095.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98095 audio codec + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - maxim,max98095 + + reg: + maxItems: 1 + + clocks: + items: + - description: master clock + + clock-names: + items: + - const: mclk + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/irq.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@11 { + compatible = "maxim,max98095"; + reg = <0x11>; + clocks = <&i2s0 0>; + clock-names = "mclk"; + }; + }; -- 2.34.1

2 1

[PATCH] ALSA: hda: Fix codec device field initializan
by Cezary Rojewski 13 Feb '23

13 Feb '23

Commit f2bd1c5ae2cb ("ALSA: hda: Fix page fault in snd_hda_codec_shutdown()") relocated initialization of several codec device fields. Due to differences between codec_exec_verb() and snd_hdac_bus_exec_bus() in how they handle VERB execution - the latter does not touch PM - assigning ->exec_verb to codec_exec_verb() causes PM to be engaged before it is configured for the device. Configuration of PM for the ASoC HDAudio sound card is done with snd_hda_set_power_save() during skl_hda_audio_probe() whereas the assignment happens early, in snd_hda_codec_device_init(). Revert to previous behavior to avoid problems caused by too early PM manipulation. Suggested-by: Jason Montleon <jmontleo(a)redhat.com> Link: https://lore.kernel.org/regressions/CALFERdzKUodLsm6=Ub3g2+PxpNpPtPq3bGBLbf… Fixes: f2bd1c5ae2cb ("ALSA: hda: Fix page fault in snd_hda_codec_shutdown()") Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- A result of recent discussion [1] regarding sound regression in v6.1 and onward. Fix made possible by outstanding input from Jason and Sasa (Cced) - Thank you! Note: I did not receive results from our CI yet, should happen on Monday but posting early so the review can be conducted. [1]: https://lore.kernel.org/regressions/CALFERdzKUodLsm6=Ub3g2+PxpNpPtPq3bGBLbf… sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index ac1cc7c5290e..2e728aad6771 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -927,7 +927,6 @@ snd_hda_codec_device_init(struct hda_bus *bus, unsigned int codec_addr, codec->depop_delay = -1; codec->fixup_id = HDA_FIXUP_ID_NOT_SET; codec->core.dev.release = snd_hda_codec_dev_release; - codec->core.exec_verb = codec_exec_verb; codec->core.type = HDA_DEV_LEGACY; mutex_init(&codec->spdif_mutex); @@ -998,6 +997,7 @@ int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card, if (snd_BUG_ON(codec_addr > HDA_MAX_CODEC_ADDRESS)) return -EINVAL; + codec->core.exec_verb = codec_exec_verb; codec->card = card; codec->addr = codec_addr; -- 2.25.1

2 2

[PATCH] ASoC: Intel: Skylake: Replace 1-element array with flex-array
by Kees Cook 13 Feb '23

13 Feb '23

The kernel is globally removing the ambiguous 0-length and 1-element arrays in favor of flexible arrays, so that we can gain both compile-time and run-time array bounds checking[1]. In this instance, struct skl_cpr_cfg contains struct skl_cpr_gtw_cfg, which defined "config_data" as a 1-element array. Normally when switching from a 1-element array to a flex-array, any related size calculations must be adjusted too. However, it seems the original code was over-allocating space, since 1 extra u32 would be included by the sizeof(): param_size = sizeof(struct skl_cpr_cfg); param_size += mconfig->formats_config[SKL_PARAM_INIT].caps_size; But the copy uses caps_size bytes, and cap_size / 4 (i.e. sizeof(u32)) for the length tracking: memcpy(cpr_mconfig->gtw_cfg.config_data, mconfig->formats_config[SKL_PARAM_INIT].caps, mconfig->formats_config[SKL_PARAM_INIT].caps_size); cpr_mconfig->gtw_cfg.config_length = (mconfig->formats_config[SKL_PARAM_INIT].caps_size) / 4; Therefore, no size calculations need adjusting. Change the struct skl_cpr_gtw_cfg config_data member to be a true flexible array, which also fixes the over-allocation, and silences this memcpy run-time false positive: memcpy: detected field-spanning write (size 100) of single field "cpr_mconfig->gtw_cfg.config_data" at sound/soc/intel/skylake/skl-messages.c:554 (size 4) [1] For lots of details, see both: https://docs.kernel.org/process/deprecated.html#zero-length-and-one-element… https://people.kernel.org/kees/bounded-flexible-arrays-in-c Reported-by: Sasa Ostrouska <casaxa(a)gmail.com> Link: https://lore.kernel.org/all/CALFERdwvq5day_sbDfiUsMSZCQu9HG8-SBpOZDNPeMdZGo… Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Cc: Bard Liao <yung-chuan.liao(a)linux.intel.com> Cc: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Cc: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: "Gustavo A. R. Silva" <gustavoars(a)kernel.org> Cc: "Amadeusz Sławiński" <amadeuszx.slawinski(a)linux.intel.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> --- sound/soc/intel/skylake/skl-topology.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index 6db0fd7bad49..ad94f8020c27 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -115,7 +115,7 @@ struct skl_cpr_gtw_cfg { u32 dma_buffer_size; u32 config_length; /* not mandatory; required only for DMIC/I2S */ - u32 config_data[1]; + u32 config_data[]; } __packed; struct skl_dma_control { -- 2.34.1

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[PATCH v4 0/4] Add CS35L41 shared boost feature
by Lucas Tanure 13 Feb '23

13 Feb '23

Valve's Steam Deck uses CS35L41 in shared boost mode, where both speakers share the boost circuit. Add this support in the shared lib, but for now, shared boost is not supported in HDA systems as would require BIOS changes. Based on David Rhodes shared boost patches. Also, fix boost config overwriting in IRQ found in the review and do a small refactor of the code. Changes from V3: - Fix wrong code sent - Fix ISO C90 mixed declarations and code Changes from V2: - Drop External boost without VSPK Documentation - Move Shared boost to use values 2 and 3 - Revert back to reg_sequence but reading the value first and only update the necessary bits - Fix bug found by Intel kernel Test Robot Changes from V1: - Fix Documentation patch subject - New patch for External boost without VSPK Documentation - New patch to fix boost IRQ overwriting issue - New patch to refactor IRQ release error code - reinit_completion on pcm_startup - fix DRE switch overwriting - return IRQ_HANDLED in PLL_LOCK case Lucas Tanure (4): ASoC: cs35l41: Only disable internal boost ASoC: cs35l41: Refactor error release code ALSA: cs35l41: Add shared boost feature ASoC: cs35l41: Document CS35l41 shared boost .../bindings/sound/cirrus,cs35l41.yaml | 10 +- include/sound/cs35l41.h | 13 +- sound/pci/hda/cs35l41_hda.c | 6 +- sound/soc/codecs/cs35l41-lib.c | 73 +++++++++- sound/soc/codecs/cs35l41.c | 125 +++++++++--------- sound/soc/codecs/cs35l41.h | 1 + 6 files changed, 157 insertions(+), 71 deletions(-) -- 2.39.1

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[PATCH V2 8/8] soundwire: amd: add pm_prepare callback and pm ops support
by Vijendar Mukunda 13 Feb '23

13 Feb '23

Add pm_prepare callback and System level pm ops support for AMD soundwire manager driver. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Signed-off-by: Mastan Katragadda <Mastan.Katragadda(a)amd.com> --- drivers/soundwire/amd_manager.c | 90 +++++++++++++++++++++++++++++++++ 1 file changed, 90 insertions(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index e53dc67f8529..350d411e695d 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -1157,6 +1157,94 @@ static int amd_sdw_clock_stop_exit(struct amd_sdw_manager *amd_manager) return 0; } +static int amd_resume_child_device(struct device *dev, void *data) +{ + int ret; + struct sdw_slave *slave = dev_to_sdw_dev(dev); + + if (!slave->probed) { + dev_dbg(dev, "skipping device, no probed driver\n"); + return 0; + } + if (!slave->dev_num_sticky) { + dev_dbg(dev, "skipping device, never detected on bus\n"); + return 0; + } + + if (!pm_runtime_suspended(dev)) + return 0; + ret = pm_request_resume(dev); + if (ret < 0) + dev_err(dev, "%s: pm_request_resume failed: %d\n", __func__, ret); + + return ret; +} + +static int __maybe_unused amd_pm_prepare(struct device *dev) +{ + struct amd_sdw_manager *amd_manager = dev_get_drvdata(dev); + struct sdw_bus *bus = &amd_manager->bus; + int ret; + + if (bus->prop.hw_disabled) { + dev_dbg(bus->dev, "Soundwire manager %d is disabled, ignoring\n", + bus->link_id); + return 0; + } + /* + * When multiple peripheral devices connected over the same link, if soundwire manager + * device is not in runtime suspend state, observed that device alerts are missing + * without pm_prepare on AMD platforms in clockstop mode0. + */ + if (pm_runtime_suspended(dev) && amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { + ret = pm_request_resume(dev); + if (ret < 0) { + dev_err(bus->dev, "pm_request_resume failed: %d\n", ret); + return 0; + } + } + /* To force peripheral devices to system level suspend state, resume the devices + * from runtime suspend state first. Without that unable to dispatch the alert + * status to peripheral driver during system level resume as they are in runtime + * suspend state. + */ + ret = device_for_each_child(bus->dev, NULL, amd_resume_child_device); + if (ret < 0) + dev_err(dev, "%s: amd_resume_child_device failed: %d\n", __func__, ret); + return 0; +} + +static int __maybe_unused amd_suspend(struct device *dev) +{ + struct amd_sdw_manager *amd_manager = dev_get_drvdata(dev); + struct sdw_bus *bus = &amd_manager->bus; + int ret; + + if (bus->prop.hw_disabled) { + dev_dbg(bus->dev, "Soundwire manager %d is disabled, ignoring\n", + bus->link_id); + return 0; + } + + if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { + ret = amd_sdw_clock_stop(amd_manager); + if (ret) + return ret; + } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { + /* + * As per hardware programming sequence on AMD platforms, + * clock stop should be invoked first before powering-off + */ + ret = amd_sdw_clock_stop(amd_manager); + if (ret) + return ret; + ret = amd_deinit_sdw_manager(amd_manager); + if (ret) + return ret; + } + return 0; +} + static int __maybe_unused amd_suspend_runtime(struct device *dev) { struct amd_sdw_manager *amd_manager = dev_get_drvdata(dev); @@ -1229,6 +1317,8 @@ static int __maybe_unused amd_resume_runtime(struct device *dev) } static const struct dev_pm_ops amd_pm = { + .prepare = amd_pm_prepare, + SET_SYSTEM_SLEEP_PM_OPS(amd_suspend, amd_resume_runtime) SET_RUNTIME_PM_OPS(amd_suspend_runtime, amd_resume_runtime, NULL) }; -- 2.34.1

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[PATCH V2 4/8] soundwire: amd: enable build for AMD soundwire manager driver
by Vijendar Mukunda 13 Feb '23

13 Feb '23

Enable build for soundwire manager driver for AMD platforms. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/Kconfig | 10 ++++++++++ drivers/soundwire/Makefile | 4 ++++ 2 files changed, 14 insertions(+) diff --git a/drivers/soundwire/Kconfig b/drivers/soundwire/Kconfig index 2b7795233282..5baf128ae60d 100644 --- a/drivers/soundwire/Kconfig +++ b/drivers/soundwire/Kconfig @@ -46,4 +46,14 @@ config SOUNDWIRE_QCOM config SOUNDWIRE_GENERIC_ALLOCATION tristate +config SOUNDWIRE_AMD + tristate "AMD Soundwire Manager driver" + select SOUNDWIRE_GENERIC_ALLOCATION + depends on ACPI && SND_SOC + help + Soundwire AMD Manager driver. + If you have an AMD platform which has a Soundwire Manager then + enable this config option to get the Soundwire support for that + device. + endif diff --git a/drivers/soundwire/Makefile b/drivers/soundwire/Makefile index ca97414ada70..5956229d3eb3 100644 --- a/drivers/soundwire/Makefile +++ b/drivers/soundwire/Makefile @@ -26,3 +26,7 @@ obj-$(CONFIG_SOUNDWIRE_INTEL) += soundwire-intel.o #Qualcomm driver soundwire-qcom-y := qcom.o obj-$(CONFIG_SOUNDWIRE_QCOM) += soundwire-qcom.o + +#AMD driver +soundwire-amd-y := amd_manager.o +obj-$(CONFIG_SOUNDWIRE_AMD) += soundwire-amd.o -- 2.34.1

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[PATCH V2 1/8] soundwire: export sdw_compute_slave_ports() function
by Vijendar Mukunda 13 Feb '23

13 Feb '23

Export sdw_compute_slave_ports() function to use it in another soundwire manager module. Move sdw_transport_data structure to bus header file to export sdw_compute_slave_ports() function. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/lkml/20230201165944.3169125-1-Vijendar.Mukunda@amd.… --- drivers/soundwire/bus.h | 9 +++++++++ drivers/soundwire/generic_bandwidth_allocation.c | 12 +++--------- 2 files changed, 12 insertions(+), 9 deletions(-) diff --git a/drivers/soundwire/bus.h b/drivers/soundwire/bus.h index 96927a143796..0ea0412842c5 100644 --- a/drivers/soundwire/bus.h +++ b/drivers/soundwire/bus.h @@ -144,6 +144,13 @@ struct sdw_master_runtime { struct list_head bus_node; }; +struct sdw_transport_data { + int hstart; + int hstop; + int block_offset; + int sub_block_offset; +}; + struct sdw_dpn_prop *sdw_get_slave_dpn_prop(struct sdw_slave *slave, enum sdw_data_direction direction, unsigned int port_num); @@ -212,5 +219,7 @@ int sdw_bwrite_no_pm_unlocked(struct sdw_bus *bus, u16 dev_num, u32 addr, u8 val void sdw_clear_slave_status(struct sdw_bus *bus, u32 request); int sdw_slave_modalias(const struct sdw_slave *slave, char *buf, size_t size); +void sdw_compute_slave_ports(struct sdw_master_runtime *m_rt, + struct sdw_transport_data *t_data); #endif /* __SDW_BUS_H */ diff --git a/drivers/soundwire/generic_bandwidth_allocation.c b/drivers/soundwire/generic_bandwidth_allocation.c index f7c66083a4dd..39543048baa7 100644 --- a/drivers/soundwire/generic_bandwidth_allocation.c +++ b/drivers/soundwire/generic_bandwidth_allocation.c @@ -28,15 +28,8 @@ struct sdw_group { unsigned int *rates; }; -struct sdw_transport_data { - int hstart; - int hstop; - int block_offset; - int sub_block_offset; -}; - -static void sdw_compute_slave_ports(struct sdw_master_runtime *m_rt, - struct sdw_transport_data *t_data) +void sdw_compute_slave_ports(struct sdw_master_runtime *m_rt, + struct sdw_transport_data *t_data) { struct sdw_slave_runtime *s_rt = NULL; struct sdw_port_runtime *p_rt; @@ -85,6 +78,7 @@ static void sdw_compute_slave_ports(struct sdw_master_runtime *m_rt, } } } +EXPORT_SYMBOL(sdw_compute_slave_ports); static void sdw_compute_master_ports(struct sdw_master_runtime *m_rt, struct sdw_group_params *params, -- 2.34.1

1 0

Fixed GPIO2 value assigned
by Kailang 13 Feb '23

13 Feb '23

Hi Takashi, This fix was GPIO2 value 0x3 change to 0x4. GPIO2 GPIO1 GPIO0 1 0 0 ==> 0x4 ==> GPIO2 HIGH GPIO2 GPIO1 GPIO0 0 1 1 ==> 0x3 ==> GPIO0 and GPIO1 HIGH BR, Kailang

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[PATCH v2 0/2] ASoC: dt-bindings: renesas,rsnd.yaml: add R-Car Gen4 support
by Kuninori Morimoto 13 Feb '23

13 Feb '23

Hi Mark, Rob, Krzysztof Cc Geert These are v2 of R-Car Gen4 support for Renesas sound driver. I have posted these patches as [RFC], because the schemas doesn't work correctly for me under certain conditions. "required: dmas/dma-names" always hits to "if-then" and never hits to "else" for some reasons. I'm still not sure why... But to be very strict, actually, this schemas is not mandatory, because SSI has PIO transfer mode which doesn't need to have DMA settings. This patch-set drops "dmas/dma-names" from SSI for now, and ignore the issue, but it is *not* a wrong schemas. v1 -> v2 - tidyup Subject - separate patches (drop "dmas/dma-names", add Gen4 support) - minItems -> maxItems - more git-logs. Link: https://lore.kernel.org/all/87zg9vk0ex.wl-kuninori.morimoto.gx@renesas.com/… Link: https://lore.kernel.org/all/87r0v2uvm7.wl-kuninori.morimoto.gx@renesas.com/… Link: https://lore.kernel.org/all/87r0v1t02h.wl-kuninori.morimoto.gx@renesas.com/… Kuninori Morimoto (2): ASoC: dt-bindings: renesas,rsnd.yaml: drop "dmas/dma-names" from "rcar_sound,ssi" ASoC: dt-bindings: renesas,rsnd.yaml: add R-Car Gen4 support .../bindings/sound/renesas,rsnd.yaml | 82 +++++++++++++++---- 1 file changed, 65 insertions(+), 17 deletions(-) -- 2.25.1

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[PATCH 0/2] Codec support on Jetson AGX Orin
by Sameer Pujar 13 Feb '23

13 Feb '23

Jetson AGX Orin has onboard RT5640 audio codec and uses I2S1 interface. Add DT bindings to enable codec support. Series also contains a clock fix in the codec driver. Sameer Pujar (2): ASoC: rt5640: Update MCLK rate in set_sysclk() arm64: tegra: Audio codec support on Jetson AGX Orin .../dts/nvidia/tegra234-p3737-0000+p3701-0000.dts | 47 +++++++++++++++++++++- sound/soc/codecs/rt5640.c | 5 +++ 2 files changed, 51 insertions(+), 1 deletion(-) -- 2.7.4

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No sound on HDMI by default on Lenovo T14s using Fedora 37
by Mike FABIAN 11 Feb '23

11 Feb '23

With a Lenovo Thinkpad T14s (Gen1, Intel version), I tried to get sound on the monitor speakers via the HDMI cable. It didn’t work out of the box and after a bit of googling I found: https://forums.lenovo.com/t5/ThinkPad-T400-T500-and-newer-T-series-Laptops/… which suggested to put snd_rn_pci_acp3x.dmic_acpi_check=0 on the kernel command line. So I tried this: $ cat /etc/modprobe.d/lenovot14s.conf options snd-rn-pci-acp3x dmic_acpi_check=0 And that did make it work indeed. I have no idea what this means and why this makes it work. -- Mike FABIAN <mfabian(a)redhat.com> 睡眠不足はいい仕事の敵だ。

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